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README.md
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# Installation from package repository
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> Each version of Asterisk is unstable and may contain many bugs.
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{.is-warning}
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|
||||
```
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||||
sudo apt update
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sudo apt install asterisk
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||||
```
|
||||
|
||||
> The package in APT is usually outdated, but rather stable.
|
||||
{.is-warning}
|
||||
|
||||
To restart the Asterisk service:
|
||||
```
|
||||
sudo systemctl restart asterisk
|
||||
```
|
||||
|
||||
To close the Asterisk service:
|
||||
```
|
||||
sudo systemctl stop asterisk
|
||||
```
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||||
|
||||
## Testing
|
||||
|
||||
To test, let's connect to the Asterisk console:
|
||||
```
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||||
sudo rasterisk
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||||
```
|
||||
|
||||
Which will bring you into the Asterisk command-line client. You will see this prompt after the basic Asterisk information is displayed:
|
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```
|
||||
asterisk*CLI>
|
||||
```
|
||||
|
||||
To change the verbosity of the console, use the following:
|
||||
```
|
||||
core set verbose 4
|
||||
```
|
||||
|
||||
To check the version of Asterisk, enter:
|
||||
```
|
||||
sudo rasterisk -V
|
||||
```
|
||||
|
||||
If any error occurs, you can see the service log or logs from files:
|
||||
```
|
||||
sudo journalctl -eu asterisk -f
|
||||
sudo vim /var/log/asterisk/full
|
||||
```
|
||||
|
||||
# pjsip.conf
|
||||
<kbd>res_pjsip</kbd> configuration is stored in <kbd>pjsip.conf</kbd>.
|
||||
|
||||
To activate PJSIP add chan_sip to noload and delete res_pjsip in <kbd>modules.conf</kbd>.
|
||||
|
||||
```
|
||||
noload => chan_sip.so
|
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;noload => res_pjsip.so
|
||||
```
|
||||
|
||||
## Intercom
|
||||
```
|
||||
[domofon]
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||||
type=endpoint
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||||
direct_media=no
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||||
rtp_symmetric=yes
|
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force_rport=yes
|
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rewrite_contact=yes
|
||||
allow=alaw,h264
|
||||
transport=transport-udp
|
||||
context=internal
|
||||
aors=domofon
|
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auth=domofon
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||||
|
||||
[domofon]
|
||||
type=auth
|
||||
auth_type=userpass
|
||||
password=your_password
|
||||
username=domofon
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|
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[domofon]
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type=aor
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max_contacts=1
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```
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### Endpoint
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| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| direct_media | determines whether media may flow directly between endpoints (default: "yes")
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||||
| rtp_symmetric | user agents (UAs) use the same socket/port binding to send and receive RTP stream packets
|
||||
| force_rport | causes Asterisk to always send responses back to the address/port from which it received requests
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| rewrite_contact | allow contact header to be rewritten with the source IP address port (default: "no")
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| allow | allow codecs in order of preference
|
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| transport | set the default transports; the order determines the primary default transport
|
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| context | default context for incoming calls (from <kbd>extensions.conf</kbd>); defaults to 'default'
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| aors | AoRs to be used with the endpoint (default: "")
|
||||
| auth | authentication objects associated with the endpoint (default: "")
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||||
|
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### Auth
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
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| auth_type | may be "userpass" for plain text passwords or "md5" for pre-hashed credentials. (default: "userpass")
|
||||
| password, username| credentials for registration
|
||||
|
||||
|
||||
### AoR
|
||||
|
||||
| Variable | Description |
|
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| ----------- | ----------- |
|
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| max_contacts | maximum number of contacts that can bind to an AoR (default: "0")
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## Client
|
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This configuration example is for registering a client application with Asterisk using WebRTC.
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|
||||
```
|
||||
[15]
|
||||
type=endpoint
|
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direct_media=no
|
||||
context=from-external
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allow=vp8,vp9,h264,alaw
|
||||
transport=transport-wss
|
||||
webrtc=yes
|
||||
auth=15
|
||||
aors=15
|
||||
|
||||
[15]
|
||||
type=auth
|
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auth_type=userpass
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||||
password=your_password
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||||
username=15
|
||||
|
||||
[15]
|
||||
type=aor
|
||||
max_contacts=10
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| webrtc | when set to "yes" this also enables the following values that are needed for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport
|
||||
|
||||
> VP8/VP9 are the default codecs for WebRTC and the only codec for intercom is H264.
|
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> In this case, a codec difference problem may occur.
|
||||
> At the time of writing only Freeswitch can transcode video from one codec to another.
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{.is-warning}
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> You can use any name for the client.
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> I chose <kbd>15</kbd> because it is convenient to call from a smartphone with a fake client (I use Zoiper).
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{.is-info}
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||||
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||||
# extensions.conf
|
||||
```
|
||||
;-------------------------------------------------------
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||||
; General Section
|
||||
;-------------------------------------------------------
|
||||
[general]
|
||||
static=yes
|
||||
writeprotect=yes
|
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clearglobalvars=yes
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autofallthrough=yes
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[from-external]
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exten => 200,1,Dial(PJSIP/domofon,120)
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||||
|
||||
[invalid]
|
||||
exten => _X,1,Stasis(hello)
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||||
same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
same => n,Hangup
|
||||
exten => _XX,1,Stasis(hello)
|
||||
same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
same => n,Hangup
|
||||
|
||||
exten => _1[0-3]X,1,Stasis(hello)
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same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
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same => n,Hangup
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||||
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||||
[default]
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||||
```
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|
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`static`
|
||||
if static is set to no, or omitted, then the pbx_config will rewrite this file when extensions are modified.
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||||
|
||||
`writeprotect`
|
||||
if static=yes and writeprotect=no, you can save dialplan by CLI command <kbd>dialplan save</kbd> too
|
||||
|
||||
`clearglobalvars`
|
||||
if clearglobalvars is set, global variables will be cleared and reparsed on a <kbd>dialplan reload</kbd>, or Asterisk reload.
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||||
|
||||
`autofallthrough`
|
||||
if autofallthrough is set, then if an extension runs out of things to do, it will terminate the call with BUSY, CONGESTION
|
||||
if autofallthrough is not set, then if an extension runs out ofthings to do, Asterisk will wait for a new extension to be dialed
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||||
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||||
`_X`, `_XX`, `_1[0-3]X`
|
||||
it means that dialplans will work with 1-139 numbers
|
||||
|
||||
`${EXTEN}`
|
||||
variable equal to dialed number
|
||||
|
||||
# WebRTC
|
||||
WebRTC requires **WSS (WebSocket Secure)** and we need a domain with a certificate. You can use a self-signed certificate, but you cannot use it with a browser and sipML5.
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||||
|
||||
## Generate certificate
|
||||
|
||||
**Let’s Encrypt** is a service offering free SSL certificates through an automated API. The most popular Let’s Encrypt client is EFF’s **Certbot**.
|
||||
|
||||
Install Certbot:
|
||||
```
|
||||
sudo add-apt-repository ppa:certbot/certbot
|
||||
sudo apt-get update
|
||||
sudo apt-get install certbot
|
||||
```
|
||||
|
||||
Certbot needs to answer a cryptographic challenge issued by the Let’s Encrypt API in order to prove we control our domain. It uses ports 80 (HTTP) or 443 (HTTPS) to accomplish this. Open up the appropriate port in your firewall:
|
||||
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||||
```
|
||||
sudo ufw allow 80
|
||||
```
|
||||
|
||||
Substitute 443 above if that’s the port you’re using. ufw will output confirmation that your rule was added:
|
||||
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||||
> Output
|
||||
> Rule added
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||||
> Rule added (v6)
|
||||
{.is-success}
|
||||
|
||||
We can now run Certbot to get our certificate. We’ll use the <kbd>--standalone</kbd> option to tell Certbot to handle the challenge using its own built-in web server. The <kbd>--preferred-challenges</kbd> option instructs Certbot to use port 80 or port 443. If you’re using port 80, you want <kbd>--preferred-challenges</kbd> http. For port 443 it would be <kbd>--preferred-challenges</kbd> tls-sni. Finally, the <kbd>-d</kbd> flag is used to specify the domain you’re requesting a certificate for. You can add multiple <kbd>-d</kbd> options to cover multiple domains in one certificate.
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||||
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||||
```
|
||||
sudo certbot certonly --standalone --preferred-challenges http -d example.com
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||||
```
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## Configure Asterisk with cerificate
|
||||
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||||
This certificate can only be accessed by the <kbd>root</kbd> user. Asterisk is executed by <kbd>asterisk</kbd> user. The way to fix this problem is to copy the certificate into Asterisk directory and change the owner.
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||||
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||||
If you use Asterisk as a Linux service:
|
||||
```
|
||||
mkdir /etc/asterisk/keys
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||||
sudo cp -L /etc/letsencrypt/live/example.com/cert.pem /etc/asterisk/keys
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||||
sudo cp -L /etc/letsencrypt/live/example.com/privkey.pem /etc/asterisk/keys
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||||
sudo chown asterisk:asterisk /etc/asterisk/keys/cert.pem
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||||
sudo chown asterisk:asterisk /etc/asterisk/keys/privkey.pem
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||||
```
|
||||
|
||||
You need to enable TLS and add the certificate to <kbd>http.conf</kbd>:
|
||||
```
|
||||
[general]
|
||||
enabled=no
|
||||
tlsenable=yes
|
||||
tlsbindaddr=0.0.0.0:8089
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||||
tlscertfile=/etc/asterisk/keys/cert.pem
|
||||
tlsprivatekey=/etc/asterisk/keys/privkey.pem
|
||||
```
|
||||
|
||||
## Testing
|
||||
|
||||
You can use one of 2 sites to test WebRTC from your browser:
|
||||
- https://www.doubango.org/sipml5/call.htm
|
||||
- https://tryit.jssip.net/
|
||||
|
||||
# HTTP server (ARI)
|
||||
The HTTP server in Asterisk is configured via http.conf. Note that this does not describe all of the options available via http.conf - rather, it lists the most useful ones for ARI.
|
||||
|
||||
Example:
|
||||
```
|
||||
[general]
|
||||
enabled = yes
|
||||
bindaddr = 0.0.0.0
|
||||
bindport = 8088
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| enabled | Enable the HTTP server. The HTTP server in Asterisk is disabled by default. Unless it is enabled, ARI will not function!
|
||||
| bindaddr | The IP address to bind the HTTP server to. This can either be an explicit local address, or 0.0.0.0 to bind to all available interfaces.
|
||||
| bindport | The port to bind the HTTP server to. Client making HTTP requests should specify 8088 as the port to send the request to.
|
||||
| prefix | A prefix to require for all requests. If specified, requests must begin with the specified prefix.
|
||||
| tlsenable | Enable HTTPS
|
||||
| tlsbindaddr | The IP address and port to bind the HTTPS server to. This should be an IP address and port, e.g., 0.0.0.0:8089
|
||||
| tlscertfile | The full path to the certificate file to use. Asterisk only supports the .pem format
|
||||
| tlsprivatekey| The full path to the private key file. Asterisk only supports the .pem format. If this is not specified, the certificate specified in tlscertfile will be searched for the private key.
|
||||
|
||||
For creating a certificate, see [WebRTC configuration](/telecom/Asterisk/webrtc)
|
||||
|
||||
# Fail2Ban
|
||||
|
||||
To install fail2ban:
|
||||
```
|
||||
sudo apt install fail2ban
|
||||
```
|
||||
|
||||
The default settings of the program are in the <kbd>/etc/fail2ban/jail.conf</kbd> file, it is recommended to change the settings in <kbd>/etc/fail2ban/jail.local</kbd>, which is a copy of <kbd>jail.conf</kbd>.
|
||||
|
||||
The file contains a section of general settings <kbd>[DEFAULT]</kbd> and sections of specific settings for certain services (for example, the presence of the <kbd>[ssh]</kbd> section is demonstrated).
|
||||
|
||||
```
|
||||
[DEFAULT]
|
||||
ignoreip = 127.0.0.1/8
|
||||
ignorecommand =
|
||||
bantime = 3600
|
||||
findtime = 600
|
||||
maxretry = 3
|
||||
backend = auto
|
||||
usedns = warn
|
||||
destemail = root@localhost
|
||||
sendername = Fail2Ban
|
||||
sender = fail2ban@localhost
|
||||
banaction = iptables-multiport
|
||||
mta = sendmail
|
||||
protocol = tcp
|
||||
chain = INPUT
|
||||
action_ = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
action_mw = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
%(mta)s-whois[name=%(__name__)s, dest="%(destemail)s", protocol="%(protocol)s", chain="%(chain)s", sendername="%(sendername)s"]
|
||||
action_mwl = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
%(mta)s-whois-lines[name=%(__name__)s, dest="%(destemail)s", logpath=%(logpath)s, chain="%(chain)s", sendername="%(sendername)s"]
|
||||
action = %(action_)s
|
||||
|
||||
[ssh]
|
||||
enabled = true
|
||||
port = ssh
|
||||
filter = sshd
|
||||
logpath = /var/log/auth.log
|
||||
maxretry = 6
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| ignoreip | can be a list of IP addresses, CIDR masks or DNS hosts. Fail2ban will not ban a host which matches an address in this list. Several addresses can be defined using space (and/or comma) separator.
|
||||
| bantime | the number of seconds that a host is banned.
|
||||
| findtime, maxretry| a host is banned if it has generated "maxretry" during the last "findtime" seconds
|
||||
| backend | specifies the backend used to get files modification
|
||||
| usedns | specifies if jails should trust hostnames in logs, warn when DNS lookups are performed, or ignore all hostnames in logsyes: if a hostname is encountered, a DNS lookup will be performed
|
||||
| destemail | destination email address used solely for the interpolations in jail.{conf,local,d/*} configuration files.
|
||||
| sender | sender email address used solely for some actions
|
||||
| banaction | default banning action (e.g. iptables, iptables-new, iptables-multiport, shorewall, etc) It is used to define a ction_* variables
|
||||
| protocol | default protocol (tcp, udp, ...)
|
||||
| chain | specify chain where jumps would need to be added in ban-actions expecting parameter chain
|
||||
| enabled | enables the jails
|
||||
| port | ports to be banned
|
||||
| filter | defines the filter to use by the jail
|
||||
| logpath | path to the logs
|
||||
|
||||
## Asterisk configuration
|
||||
|
||||
1. Copy the configuration file to the local one.
|
||||
```
|
||||
sudo cp /etc/fail2ban/jail.conf /etc/fail2ban/jail.local
|
||||
```
|
||||
|
||||
2. Add to <kbd>ignoreip</kbd> the addresses from which you are going to connect to the asterisk.
|
||||
|
||||
3. Add Asterisk jail (see below). You can set up variables however you like.
|
||||
|
||||
```
|
||||
[asterisk]
|
||||
enabled = true
|
||||
filter = asterisk
|
||||
backend = auto
|
||||
port = 5060,5061
|
||||
action = iptables-allports[name=ASTERISK, protocol=all, blocktype=DROP]
|
||||
logpath = /var/log/asterisk/messages
|
||||
findtime = 1m
|
||||
maxretry = 5
|
||||
bantime = 30d
|
||||
```
|
||||
|
||||
4. (optional) If you have ssh, it is more secure if you add it to the jail.
|
||||
|
||||
```
|
||||
[sshd]
|
||||
enabled = true
|
||||
bantime = 60m
|
||||
findtime = 1m
|
||||
maxretry = 5
|
||||
port = ssh
|
||||
logpath = %(sshd_log)s
|
||||
backend = %(sshd_backend)s
|
||||
```
|
||||
80
acl.conf
Executable file
80
acl.conf
Executable file
@ -0,0 +1,80 @@
|
||||
;
|
||||
; Named Access Control Lists (ACLs)
|
||||
;
|
||||
; A convenient way to share acl definitions
|
||||
;
|
||||
; This configuration file is read on startup
|
||||
;
|
||||
; CLI Commands
|
||||
; -----------------------------------------------------------
|
||||
; acl show Show all named ACLs configured
|
||||
; acl show <name> Show contents of a particular named ACL
|
||||
; reload acl Reload configuration file
|
||||
;
|
||||
; Any configuration that uses ACLs which has been made to be able to use named
|
||||
; ACLs will specify a named ACL with the 'acl' option in its configuration in
|
||||
; a similar fashion to the usual 'permit' and 'deny' options. Example:
|
||||
; acl=my_named_acl
|
||||
;
|
||||
; Multiple named ACLs can be applied by either comma separating the arguments or
|
||||
; just by adding additional ACL lines. Example:
|
||||
; acl=my_named_acl
|
||||
; acl=my_named_acl2
|
||||
;
|
||||
; or
|
||||
;
|
||||
; acl=my_named_acl,my_named_acl2
|
||||
;
|
||||
; ACLs specified by name are evaluated independently from the ACL specified via
|
||||
; permit/deny. In order for an address to pass a given ACL, it must pass both
|
||||
; the ACL specified by permit/deny for a given item as well as any named ACLs
|
||||
; that were specified.
|
||||
;
|
||||
;[example_named_acl1]
|
||||
;deny=0.0.0.0/0.0.0.0
|
||||
;permit=209.16.236.0
|
||||
;permit=209.16.236.1
|
||||
;
|
||||
;[example_named_acl2]
|
||||
;permit=0.0.0.0/0.0.0.0
|
||||
;deny=10.24.20.171
|
||||
;deny=10.24.20.103
|
||||
;deny=209.16.236.1
|
||||
;
|
||||
; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
|
||||
; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
|
||||
; The following example explains how combining the ACLs works:
|
||||
; <in another configuration>
|
||||
; [example_item_with_acl]
|
||||
; acl=example_named_acl1
|
||||
; acl=example_named_acl2
|
||||
;
|
||||
; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
|
||||
; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
|
||||
; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
|
||||
; either, so it passes.
|
||||
;
|
||||
; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
|
||||
; First, example_named_acl1 is evaluated and the address is allowed.
|
||||
; However, it is blocked by example_named_acl2, so the address is blocked from the combined
|
||||
; ACL.
|
||||
;
|
||||
; Similarly, the permits/denies in specific configurations that make up an ACL definition
|
||||
; are also treated as a separate ACL for evaluation. So if we change the example above to:
|
||||
; <in another configuration>
|
||||
; [example_item_with_acl]
|
||||
; acl=example_named_acl1
|
||||
; acl=example_named_acl2
|
||||
; deny=209.16.236.0
|
||||
;
|
||||
; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
|
||||
; though it passes the two named components.
|
||||
;
|
||||
;
|
||||
; Named ACLs can use ipv6 addresses just like normal ACLs.
|
||||
;[ipv6_example_1]
|
||||
;deny = ::
|
||||
;permit = ::1/128
|
||||
;
|
||||
;[ipv6_example_2]
|
||||
;permit = fe80::21d:bad:fad:2323
|
||||
8
adsi.conf
Executable file
8
adsi.conf
Executable file
@ -0,0 +1,8 @@
|
||||
;
|
||||
; Sample ADSI Configuration file
|
||||
;
|
||||
[intro]
|
||||
alignment = center
|
||||
greeting => Welcome to the
|
||||
greeting => Asterisk
|
||||
greeting => Open Source PBX
|
||||
70
agents.conf
Executable file
70
agents.conf
Executable file
@ -0,0 +1,70 @@
|
||||
;
|
||||
; Agent pool configuration
|
||||
;
|
||||
|
||||
[general]
|
||||
; The general section of this config is not currently used, but reserved
|
||||
; for future use.
|
||||
|
||||
;[agent-id]
|
||||
; Define ackcall to require the agent to give a DTMF acknowledgement
|
||||
; when the agent receives a call.
|
||||
; The channel variable AGENTACKCALL overrides on agent login.
|
||||
; Default is "no".
|
||||
;ackcall=no
|
||||
;
|
||||
; Set what DTMF key sequence the agent should use to acknowledge a call.
|
||||
; The channel variable AGENTACCEPTDTMF overrides on agent login.
|
||||
; This option is ignored unless ackcall is enabled.
|
||||
; Default is "#".
|
||||
;acceptdtmf=##
|
||||
;
|
||||
; Set how many seconds a call for the agent has to wait for the agent to
|
||||
; acknowledge the call before the agent is automatically logged off. If
|
||||
; set to zero then the call will wait forever for the agent to acknowledge.
|
||||
; The channel variable AGENTAUTOLOGOFF overrides on agent login.
|
||||
; This option is ignored unless ackcall is enabled.
|
||||
; Default is 0.
|
||||
;autologoff=15
|
||||
;
|
||||
; Set the minimum amount of time after disconnecting a call before
|
||||
; the agent can receive a new call in milliseconds.
|
||||
; The channel variable AGENTWRAPUPTIME overrides on agent login.
|
||||
; Default is 0.
|
||||
;wrapuptime=5000
|
||||
;
|
||||
; Set the musiconhold class for the agent.
|
||||
; Default is "default".
|
||||
;musiconhold=default
|
||||
;
|
||||
; Enable recording calls the agent takes automatically by invoking the
|
||||
; DTMF automixmon feature when the agent connects to a caller.
|
||||
; See features.conf.sample for information about the automixmon feature.
|
||||
; Default is "no".
|
||||
;recordagentcalls=yes
|
||||
;
|
||||
; The sound file played to alert the agent when a call is present.
|
||||
; Default is "beep".
|
||||
;custom_beep=beep
|
||||
;
|
||||
; A friendly name for the agent used in log messages.
|
||||
; Default is "".
|
||||
;fullname=Mark Spencer
|
||||
;
|
||||
; --------------------------------------------------
|
||||
;
|
||||
; This section contains example agent definitions:
|
||||
;
|
||||
; Define a template called my-agents:
|
||||
;[my-agents](!)
|
||||
;autologoff=15
|
||||
;ackcall=yes
|
||||
;acceptdtmf=##
|
||||
;
|
||||
; Define agent 1001 using the my-agents template:
|
||||
;[1001](my-agents)
|
||||
;fullname=Mark Spencer
|
||||
;
|
||||
; Define agent 1002 using the my-agents template:
|
||||
;[1002](my-agents)
|
||||
;fullname=Will Meadows
|
||||
91
alarmreceiver.conf
Executable file
91
alarmreceiver.conf
Executable file
@ -0,0 +1,91 @@
|
||||
;
|
||||
; alarmreceiver.conf
|
||||
;
|
||||
; Sample configuration file for the Asterisk alarm receiver application.
|
||||
;
|
||||
|
||||
|
||||
[general]
|
||||
|
||||
;
|
||||
; Specify a timestamp format for the metadata section of the event files
|
||||
; Default is %a %b %d, %Y @ %H:%M:%S %Z
|
||||
|
||||
timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
|
||||
|
||||
;
|
||||
; Specify a command to execute when the caller hangs up
|
||||
;
|
||||
; Default is none
|
||||
;
|
||||
|
||||
;eventcmd = yourprogram -yourargs ...
|
||||
|
||||
;
|
||||
; Specify a spool directory for the event files. This setting is required
|
||||
; if you want the app to be useful. Event files written to the spool
|
||||
; directory will be of the template event-XXXXXX, where XXXXXX is a random
|
||||
; and unique alphanumeric string.
|
||||
;
|
||||
; Default is none, and the events will be dropped on the floor.
|
||||
;
|
||||
|
||||
eventspooldir = /tmp
|
||||
|
||||
;
|
||||
; The alarmreceiver app can either log the events one-at-a-time to individual
|
||||
; files in the spool directory, or it can store them until the caller
|
||||
; disconnects and write them all to one file.
|
||||
;
|
||||
; The default setting for logindividualevents is no.
|
||||
;
|
||||
|
||||
logindividualevents = no
|
||||
|
||||
;
|
||||
; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
|
||||
; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
|
||||
; receiver by entering digits manually, set this to a reasonable time out
|
||||
; like 10000 milliseconds.
|
||||
|
||||
fdtimeout = 2000
|
||||
|
||||
;
|
||||
; The timeout for receiving subsequent DTMF digits is adjustable from
|
||||
; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
|
||||
; the receiver by entering digits manually, set this to a reasonable time out
|
||||
; like 4000 milliseconds.
|
||||
;
|
||||
|
||||
sdtimeout = 200
|
||||
|
||||
;
|
||||
; Wait for the connection to settle post-answer. Adjustable from 500 msec. to 10000 msec.
|
||||
; The default is 1250 msec.
|
||||
;
|
||||
|
||||
answait = 1250
|
||||
|
||||
; When logging individual events it may be desirable to skip grouping of metadata
|
||||
|
||||
;no_group_meta = yes
|
||||
|
||||
;
|
||||
; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
|
||||
; The default is 8192. This shouldn't need to be messed with, but is included
|
||||
; just in case there are problems with signal levels.
|
||||
;
|
||||
|
||||
loudness = 8192
|
||||
|
||||
;
|
||||
; The db-family setting allows the user to capture statistics on the number of
|
||||
; calls, and the errors the alarm receiver sees. The default is for no
|
||||
; db-family name to be defined and the database logging to be turned off.
|
||||
;
|
||||
|
||||
;db-family = yourfamily:
|
||||
|
||||
;
|
||||
; End of alarmreceiver.conf
|
||||
;
|
||||
77
alsa.conf
Executable file
77
alsa.conf
Executable file
@ -0,0 +1,77 @@
|
||||
;
|
||||
; Open Sound System Console Driver Configuration File
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; Automatically answer incoming calls on the console? Choose yes if
|
||||
; for example you want to use this as an intercom.
|
||||
;
|
||||
autoanswer=yes
|
||||
;
|
||||
; Default context (is overridden with @context syntax)
|
||||
;
|
||||
context=local
|
||||
;
|
||||
; Default extension to call
|
||||
;
|
||||
extension=s
|
||||
;
|
||||
; Default language
|
||||
;
|
||||
;language=en
|
||||
;
|
||||
; Default Music on Hold class to use when this channel is placed on hold in
|
||||
; the case that the music class is not set on the channel with
|
||||
; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
|
||||
; putting this one on hold did not suggest a class to use.
|
||||
;
|
||||
;mohinterpret=default
|
||||
;
|
||||
; Silence suppression can be enabled when sound is over a certain threshold.
|
||||
; The value for the threshold should probably be between 500 and 2000 or so,
|
||||
; but your mileage may vary. Use the echo test to evaluate the best setting.
|
||||
;silencesuppression = yes
|
||||
;silencethreshold = 1000
|
||||
;
|
||||
; To set which ALSA device to use, change this parameter
|
||||
;input_device=hw:0,0
|
||||
;output_device=hw:0,0
|
||||
|
||||
;
|
||||
; Default mute state (can also be toggled via CLI)
|
||||
;mute=true
|
||||
|
||||
;
|
||||
; If enabled, no audio capture device will be opened. This is useful on
|
||||
; systems where there will be no return audio path, such as overhead pagers.
|
||||
;noaudiocapture=true
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The ALSA channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive ALSA side will always
|
||||
; be used if the sending side can create jitter.
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
27
amd.conf
Executable file
27
amd.conf
Executable file
@ -0,0 +1,27 @@
|
||||
;
|
||||
; Answering Machine Detection Configuration
|
||||
;
|
||||
|
||||
[general]
|
||||
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
|
||||
; on whether the audio represents a HUMAN, or a MACHINE
|
||||
silence_threshold = 256 ; If the average level of noise in a sample does not reach
|
||||
; this value, from a scale of 0 to 32767, then we will consider
|
||||
; it to be silence.
|
||||
|
||||
; Greeting ;
|
||||
initial_silence = 2500 ; Maximum silence duration before the greeting.
|
||||
; If exceeded, then the result is detection as a MACHINE.
|
||||
after_greeting_silence = 800 ; Silence after detecting a greeting.
|
||||
; If exceeded, then the result is detection as a HUMAN
|
||||
greeting = 1500 ; Maximum length of a greeting. If exceeded, then the
|
||||
; result is detection as a MACHINE.
|
||||
|
||||
; Word detection ;
|
||||
min_word_length = 100 ; Minimum duration of Voice to considered as a word
|
||||
maximum_word_length = 5000 ; Maximum duration of a single Voice utterance allowed.
|
||||
between_words_silence = 50 ; Minimum duration of silence after a word to consider
|
||||
; the audio what follows as a new word
|
||||
|
||||
maximum_number_of_words = 2 ; Maximum number of words in the greeting
|
||||
; If exceeded, then the result is detection as a MACHINE
|
||||
24
app_mysql.conf
Executable file
24
app_mysql.conf
Executable file
@ -0,0 +1,24 @@
|
||||
; Configuration file for the MYSQL app addon
|
||||
|
||||
[general]
|
||||
;
|
||||
; Nullvalue governs how NULL values are returned from the database. In
|
||||
; previous versions, the special NULL value was returned as the "NULL"
|
||||
; string. We now provide an option for the behavior, configured globally.
|
||||
; nullstring - the string "NULL"
|
||||
; emptystring - the string ""
|
||||
; null - unset the variable
|
||||
;
|
||||
; WARNING: setting nullvalue=null may have undesireable consequences, in
|
||||
; particular if you use subroutines in AEL or the LOCAL() variable construct.
|
||||
; You have been warned. Don't complain if you use that setting in combination
|
||||
; with Gosub or AEL and get buggy behavior.
|
||||
;
|
||||
nullvalue = nullstring
|
||||
|
||||
; If set, autoclear will destroy allocated statement and connection resources
|
||||
; when the channel ends. For most usage of the MYSQL app, this is what you
|
||||
; want, but it's conceivable that somebody is sharing MYSQL connections across
|
||||
; multiple channels, in which case, this should be set to 'no'. Defaults to
|
||||
; 'no', as this was the original behavior.
|
||||
autoclear=yes
|
||||
27
app_skel.conf
Executable file
27
app_skel.conf
Executable file
@ -0,0 +1,27 @@
|
||||
[general]
|
||||
games=3
|
||||
cheat=no
|
||||
|
||||
[sounds]
|
||||
prompt=please-enter-your,number,queue-less-than
|
||||
wrong_guess=vm-pls-try-again
|
||||
right_guess=auth-thankyou
|
||||
too_high=high
|
||||
too_low=low
|
||||
lose=vm-goodbye
|
||||
|
||||
[easy]
|
||||
max_number=10
|
||||
max_guesses=4
|
||||
|
||||
[medium]
|
||||
max_number=100
|
||||
max_guesses=6
|
||||
|
||||
[hard]
|
||||
max_number=1000
|
||||
max_guesses=7
|
||||
|
||||
[nightmare]
|
||||
max_number=1000
|
||||
max_guesses=1
|
||||
15
ari.conf
Executable file
15
ari.conf
Executable file
@ -0,0 +1,15 @@
|
||||
[general]
|
||||
enabled=yes
|
||||
;pretty=yes ; we don't need pretty-printing of the JSON responses in this
|
||||
; example, but you might if you use curl a lot.
|
||||
;
|
||||
; In this example, we are going to use the version of Swagger-UI that is hosted
|
||||
; at ari.asterisk.org. In order to get past CORS restrictions in the browser,
|
||||
; That origin needs to be added to the allowed_origins list.
|
||||
;
|
||||
allowed_origins=http://ari.asterisk.org
|
||||
|
||||
[hey]
|
||||
type=user
|
||||
password=jvehDRCWRwWQCGdF
|
||||
;read_only=no ; Set to yes for read-only applications
|
||||
37
ari.conf.default
Executable file
37
ari.conf.default
Executable file
@ -0,0 +1,37 @@
|
||||
[general]
|
||||
enabled = yes ; When set to no, ARI support is disabled.
|
||||
;pretty = no ; When set to yes, responses from ARI are
|
||||
; ; formatted to be human readable.
|
||||
;allowed_origins = ; Comma separated list of allowed origins, for
|
||||
; ; Cross-Origin Resource Sharing. May be set to * to
|
||||
; ; allow all origins.
|
||||
;auth_realm = ; Realm to use for authentication. Defaults to Asterisk
|
||||
; ; REST Interface.
|
||||
;
|
||||
; Default write timeout to set on websockets. This value may need to be adjusted
|
||||
; for connections where Asterisk must write a substantial amount of data and the
|
||||
; receiving clients are slow to process the received information. Value is in
|
||||
; milliseconds; default is 100 ms.
|
||||
;websocket_write_timeout = 100
|
||||
;
|
||||
; Display certain channel variables every time a channel-oriented
|
||||
; event is emitted:
|
||||
;
|
||||
; Note that this does incur a performance penalty and should be avoided if possible.
|
||||
;
|
||||
;channelvars = var1,var2,var3
|
||||
|
||||
;[username]
|
||||
;type = user ; Specifies user configuration
|
||||
;read_only = no ; When set to yes, user is only authorized for
|
||||
; ; read-only requests.
|
||||
;
|
||||
;password = ; Crypted or plaintext password (see password_format).
|
||||
;
|
||||
; password_format may be set to plain (the default) or crypt. When set to crypt,
|
||||
; crypt(3) is used to validate the password. A crypted password can be generated
|
||||
; using mkpasswd -m sha-512.
|
||||
;
|
||||
; When set to plain, the password is in plaintext.
|
||||
;
|
||||
;password_format = plain
|
||||
63
ast_debug_tools.conf
Executable file
63
ast_debug_tools.conf
Executable file
@ -0,0 +1,63 @@
|
||||
#
|
||||
# This file is used by the Asterisk debug tools.
|
||||
# Unlike other Asterisk config files, this one is
|
||||
# "sourced" by bash and must adhere to bash semantics.
|
||||
#
|
||||
|
||||
# A list of coredumps and/or coredump search patterns.
|
||||
# Bash extended globs are enabled and any resulting files
|
||||
# that aren't actually coredumps are silently ignored
|
||||
# so you can be liberal with the globs.
|
||||
#
|
||||
# If your patterns contains spaces be sure to only quote
|
||||
# the portion of the pattern that DOESN'T contain wildcard
|
||||
# expressions. If you quote the whole pattern, it won't
|
||||
# be expanded and the glob characters will be treated as
|
||||
# literals.
|
||||
#
|
||||
# The exclusion of files ending ".txt" is just for
|
||||
# demonstration purposes as non-coredumps will be ignored
|
||||
# anyway.
|
||||
COREDUMPS=(/tmp/core[-._]asterisk!(*.txt) /tmp/core[-._]$(hostname)!(*.txt))
|
||||
|
||||
# The directory to contain output files and work directories.
|
||||
# For output from existing core files, the default is the
|
||||
# directory that the core file is found in. For core files
|
||||
# produced from a running process, the default is /tmp.
|
||||
OUTPUTDIR=/tmp
|
||||
|
||||
# Date command for the "running" coredump and tarballs.
|
||||
# DATEFORMAT will be executed to get the timestamp.
|
||||
# Don't put quotes around the format string or they'll be
|
||||
# treated as literal characters. Also be aware of colons
|
||||
# in the output as you can't upload files with colons in
|
||||
# the name to Jira.
|
||||
#
|
||||
# Unix timestamp
|
||||
#DATEFORMAT='date +%s.%N'
|
||||
#
|
||||
# Unix timestamp on *BSD/MacOS after installing coreutils
|
||||
#DATEFORMAT='gdate +%s.%N'
|
||||
#
|
||||
# Readable GMT
|
||||
#DATEFORMAT='date -u +%FT%H-%M-%S%z'
|
||||
#
|
||||
# Readable Local time
|
||||
DATEFORMAT='date +%FT%H-%M-%S%z'
|
||||
|
||||
# A list of log files and/or log file search patterns using the
|
||||
# same syntax as COREDUMPS.
|
||||
#
|
||||
LOGFILES=(/var/log/asterisk/messages* /var/log/asterisk/queue* \
|
||||
/var/log/asterisk/debug* /var/log/asterisk/security*)
|
||||
|
||||
# ast_loggrabber converts POSIX timestamps to readable format
|
||||
# using this Python strftime format string. If not specified
|
||||
# or an empty string, no format covnersion is done.
|
||||
LOG_DATEFORMAT="%m/%d-%H:%M:%S.%f"
|
||||
|
||||
# The timezone to use when converting POSIX timestamps to
|
||||
# readable format. It can be specified in "<continent>/<city>"
|
||||
# format or in abbreviation format such as "CST6CDT". If not
|
||||
# specified, the "local" timezone is used.
|
||||
# LOG_TIMEZONE=
|
||||
158
asterisk.adsi
Executable file
158
asterisk.adsi
Executable file
@ -0,0 +1,158 @@
|
||||
;
|
||||
; Asterisk default ADSI script
|
||||
;
|
||||
;
|
||||
; Begin with the preamble requirements
|
||||
;
|
||||
DESCRIPTION "Asterisk PBX" ; Name of vendor
|
||||
VERSION 0x00 ; Version of stuff
|
||||
;SECURITY "_AST" ; Security code
|
||||
SECURITY 0X9BDBF7AC ; Security code
|
||||
FDN 0x0000000F ; Descriptor number
|
||||
|
||||
;
|
||||
; Flags
|
||||
;
|
||||
FLAG "nocallwaiting"
|
||||
|
||||
;
|
||||
; Predefined strings
|
||||
;
|
||||
DISPLAY "titles" IS "** Asterisk PBX **"
|
||||
DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
|
||||
DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
|
||||
DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
|
||||
DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
|
||||
DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
|
||||
DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
|
||||
DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
|
||||
DISPLAY "busy" IS "Busy." JUSTIFY LEFT
|
||||
DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
|
||||
DISPLAY "cwdisabled" IS "Callwait disabled"
|
||||
DISPLAY "empty" IS "asdf"
|
||||
|
||||
;
|
||||
; Begin soft key definitions
|
||||
;
|
||||
KEY "callfwd" IS "CallFwd" OR "Call Forward"
|
||||
OFFHOOK
|
||||
VOICEMODE
|
||||
WAITDIALTONE
|
||||
SENDDTMF "*60"
|
||||
GOTO "offHook"
|
||||
ENDKEY
|
||||
|
||||
KEY "vmail_OH" IS "VMail" OR "Voicemail"
|
||||
OFFHOOK
|
||||
VOICEMODE
|
||||
WAITDIALTONE
|
||||
SENDDTMF "8500"
|
||||
ENDKEY
|
||||
|
||||
KEY "vmail" IS "VMail" OR "Voicemail"
|
||||
SENDDTMF "8500"
|
||||
ENDKEY
|
||||
|
||||
KEY "backspace" IS "BackSpc" OR "Backspace"
|
||||
BACKSPACE
|
||||
ENDKEY
|
||||
|
||||
KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
|
||||
SENDDTMF "*70"
|
||||
SETFLAG "nocallwaiting"
|
||||
SHOWDISPLAY "cwdisabled" AT 4
|
||||
TIMERCLEAR
|
||||
TIMERSTART 1
|
||||
ENDKEY
|
||||
|
||||
KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
|
||||
SENDDTMF "*67"
|
||||
SETFLAG "nocallwaiting"
|
||||
ENDKEY
|
||||
|
||||
;
|
||||
; Begin main subroutine
|
||||
;
|
||||
|
||||
SUB "main" IS
|
||||
IFEVENT NEARANSWER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "talkingto" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" AT 3
|
||||
SHOWDISPLAY "callnum" AT 4
|
||||
GOTO "stableCall"
|
||||
ENDIF
|
||||
IFEVENT OFFHOOK THEN
|
||||
CLEAR
|
||||
CLEARFLAG "nocallwaiting"
|
||||
CLEARDISPLAY
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail"
|
||||
SHOWKEYS "cidblock"
|
||||
SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
|
||||
GOTO "offHook"
|
||||
ENDIF
|
||||
IFEVENT IDLE THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail_OH"
|
||||
ENDIF
|
||||
IFEVENT CALLERID THEN
|
||||
CLEAR
|
||||
; SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
; SHOWDISPLAY "incoming" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" AT 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" AT 4
|
||||
ENDIF
|
||||
IFEVENT RING THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "incoming" AT 2
|
||||
ENDIF
|
||||
IFEVENT ENDOFRING THEN
|
||||
SHOWDISPLAY "missedcall" AT 2
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail_OH"
|
||||
ENDIF
|
||||
IFEVENT TIMER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "empty" AT 4
|
||||
ENDIF
|
||||
ENDSUB
|
||||
|
||||
SUB "offHook" IS
|
||||
IFEVENT FARRING THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "ringing" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
IFEVENT FARANSWER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "talkingto" AT 2
|
||||
GOTO "stableCall"
|
||||
ENDIF
|
||||
IFEVENT BUSY THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "busy" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
IFEVENT REORDER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "reorder" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
ENDSUB
|
||||
|
||||
SUB "stableCall" IS
|
||||
IFEVENT REORDER THEN
|
||||
SHOWDISPLAY "callended" AT 2
|
||||
ENDIF
|
||||
ENDSUB
|
||||
133
asterisk.conf
Executable file
133
asterisk.conf
Executable file
@ -0,0 +1,133 @@
|
||||
[directories](!)
|
||||
astetcdir => /etc/asterisk
|
||||
astmoddir => /usr/lib/asterisk/modules
|
||||
astvarlibdir => /var/lib/asterisk
|
||||
astdbdir => /var/lib/asterisk
|
||||
astkeydir => /var/lib/asterisk
|
||||
astdatadir => /var/lib/asterisk
|
||||
astagidir => /var/lib/asterisk/agi-bin
|
||||
astspooldir => /var/spool/asterisk
|
||||
astrundir => /var/run/asterisk
|
||||
astlogdir => /var/log/asterisk
|
||||
astsbindir => /usr/sbin
|
||||
|
||||
[options]
|
||||
verbose = 3
|
||||
debug = 3
|
||||
;trace = 0 ; Set the trace level.
|
||||
;refdebug = yes ; Enable reference count debug logging.
|
||||
;alwaysfork = yes ; Same as -F at startup.
|
||||
;nofork = yes ; Same as -f at startup.
|
||||
;quiet = yes ; Same as -q at startup.
|
||||
;timestamp = yes ; Same as -T at startup.
|
||||
;execincludes = yes ; Support #exec in config files.
|
||||
;console = yes ; Run as console (same as -c at startup).
|
||||
;highpriority = yes ; Run realtime priority (same as -p at
|
||||
; startup).
|
||||
;initcrypto = yes ; Initialize crypto keys (same as -i at
|
||||
; startup).
|
||||
;nocolor = yes ; Disable console colors.
|
||||
;dontwarn = yes ; Disable some warnings.
|
||||
;dumpcore = yes ; Dump core on crash (same as -g at startup).
|
||||
;languageprefix = yes ; Use the new sound prefix path syntax.
|
||||
;systemname = my_system_name ; Prefix uniqueid with a system name for
|
||||
; Global uniqueness issues.
|
||||
;autosystemname = yes ; Automatically set systemname to hostname,
|
||||
; uses 'localhost' on failure, or systemname if
|
||||
; set.
|
||||
;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
|
||||
; If we get shorter DTMF messages, these will be
|
||||
; changed to the minimum duration
|
||||
;maxcalls = 10 ; Maximum amount of calls allowed.
|
||||
;maxload = 0.9 ; Asterisk stops accepting new calls if the
|
||||
; load average exceed this limit.
|
||||
;maxfiles = 1000 ; Maximum amount of openfiles.
|
||||
;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if
|
||||
; the amount of free memory falls below this
|
||||
; watermark.
|
||||
;cache_media_frames = yes ; Cache media frames for performance
|
||||
; Disable this option to help track down media frame
|
||||
; mismanagement when using valgrind or MALLOC_DEBUG.
|
||||
; The cache gets in the way of determining if the
|
||||
; frame is used after being freed and who freed it.
|
||||
; NOTE: This option has no effect when Asterisk is
|
||||
; compiled with the LOW_MEMORY compile time option
|
||||
; enabled because the cache code does not exist.
|
||||
; Default yes
|
||||
;cache_record_files = yes ; Cache recorded sound files to another
|
||||
; directory during recording.
|
||||
;record_cache_dir = /tmp ; Specify cache directory (used in conjunction
|
||||
; with cache_record_files).
|
||||
;transmit_silence = yes ; Transmit silence while a channel is in a
|
||||
; waiting state, a recording only state, or
|
||||
; when DTMF is being generated. Note that the
|
||||
; silence internally is generated in raw signed
|
||||
; linear format. This means that it must be
|
||||
; transcoded into the native format of the
|
||||
; channel before it can be sent to the device.
|
||||
; It is for this reason that this is optional,
|
||||
; as it may result in requiring a temporary
|
||||
; codec translation path for a channel that may
|
||||
; not otherwise require one.
|
||||
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
|
||||
; directly.
|
||||
;runuser = asterisk ; The user to run as.
|
||||
;rungroup = asterisk ; The group to run as.
|
||||
;lightbackground = yes ; If your terminal is set for a light-colored
|
||||
; background.
|
||||
;forceblackbackground = yes ; Force the background of the terminal to be
|
||||
; black, in order for terminal colors to show
|
||||
; up properly.
|
||||
;defaultlanguage = en ; Default language
|
||||
documentation_language = en_US ; Set the language you want documentation
|
||||
; displayed in. Value is in the same format as
|
||||
; locale names.
|
||||
;hideconnect = yes ; Hide messages displayed when a remote console
|
||||
; connects and disconnects.
|
||||
;lockconfdir = no ; Protect the directory containing the
|
||||
; configuration files (/etc/asterisk) with a
|
||||
; lock.
|
||||
;stdexten = gosub ; How to invoke the extensions.conf stdexten.
|
||||
; macro - Invoke the stdexten using a macro as
|
||||
; done by legacy Asterisk versions.
|
||||
; gosub - Invoke the stdexten using a gosub as
|
||||
; documented in extensions.conf.sample.
|
||||
; Default gosub.
|
||||
;live_dangerously = no ; Enable the execution of 'dangerous' dialplan
|
||||
; functions from external sources (AMI,
|
||||
; etc.) These functions (such as SHELL) are
|
||||
; considered dangerous because they can allow
|
||||
; privilege escalation.
|
||||
; Default no
|
||||
;entityid=00:11:22:33:44:55 ; Entity ID.
|
||||
; This is in the form of a MAC address.
|
||||
; It should be universally unique.
|
||||
; It must be unique between servers communicating
|
||||
; with a protocol that uses this value.
|
||||
; This is currently is used by DUNDi and
|
||||
; Exchanging Device and Mailbox State
|
||||
; using protocols: XMPP, Corosync and PJSIP.
|
||||
;rtp_use_dynamic = yes ; When set to "yes" RTP dynamic payload types
|
||||
; are assigned dynamically per RTP instance vs.
|
||||
; allowing Asterisk to globally initialize them
|
||||
; to pre-designated numbers (defaults to "yes").
|
||||
;rtp_pt_dynamic = 35 ; Normally the Dynamic RTP Payload Type numbers
|
||||
; are 96-127, which allow just 32 formats. The
|
||||
; starting point 35 enables the range 35-63 and
|
||||
; allows 29 additional formats. When you use
|
||||
; more than 32 formats in the dynamic range and
|
||||
; calls are not accepted by a remote
|
||||
; implementation, please report this and go
|
||||
; back to value 96.
|
||||
;hide_messaging_ami_events = no; This option, if enabled, will
|
||||
; suppress all of the Message/ast_msg_queue channel's
|
||||
; housekeeping AMI and ARI channel events. This can
|
||||
; reduce the load on the manager and ARI applications
|
||||
; when the Digium Phone Module for Asterisk is in use.
|
||||
|
||||
; Changing the following lines may compromise your security.
|
||||
;[files]
|
||||
;astctlpermissions = 0660
|
||||
;astctlowner = root
|
||||
;astctlgroup = apache
|
||||
;astctl = asterisk.ctl
|
||||
BIN
asterisk/.pjsip.conf.swp
Executable file
BIN
asterisk/.pjsip.conf.swp
Executable file
Binary file not shown.
381
asterisk/README.md
Executable file
381
asterisk/README.md
Executable file
@ -0,0 +1,381 @@
|
||||
# Installation from package repository
|
||||
|
||||
> Each version of Asterisk is unstable and may contain many bugs.
|
||||
{.is-warning}
|
||||
|
||||
```
|
||||
sudo apt update
|
||||
sudo apt install asterisk
|
||||
```
|
||||
|
||||
> The package in APT is usually outdated, but rather stable.
|
||||
{.is-warning}
|
||||
|
||||
To restart the Asterisk service:
|
||||
```
|
||||
sudo systemctl restart asterisk
|
||||
```
|
||||
|
||||
To close the Asterisk service:
|
||||
```
|
||||
sudo systemctl stop asterisk
|
||||
```
|
||||
|
||||
## Testing
|
||||
|
||||
To test, let's connect to the Asterisk console:
|
||||
```
|
||||
sudo rasterisk
|
||||
```
|
||||
|
||||
Which will bring you into the Asterisk command-line client. You will see this prompt after the basic Asterisk information is displayed:
|
||||
```
|
||||
asterisk*CLI>
|
||||
```
|
||||
|
||||
To change the verbosity of the console, use the following:
|
||||
```
|
||||
core set verbose 4
|
||||
```
|
||||
|
||||
To check the version of Asterisk, enter:
|
||||
```
|
||||
sudo rasterisk -V
|
||||
```
|
||||
|
||||
If any error occurs, you can see the service log or logs from files:
|
||||
```
|
||||
sudo journalctl -eu asterisk -f
|
||||
sudo vim /var/log/asterisk/full
|
||||
```
|
||||
|
||||
# pjsip.conf
|
||||
<kbd>res_pjsip</kbd> configuration is stored in <kbd>pjsip.conf</kbd>.
|
||||
|
||||
To activate PJSIP add chan_sip to noload and delete res_pjsip in <kbd>modules.conf</kbd>.
|
||||
|
||||
```
|
||||
noload => chan_sip.so
|
||||
;noload => res_pjsip.so
|
||||
```
|
||||
|
||||
## Intercom
|
||||
```
|
||||
[domofon]
|
||||
type=endpoint
|
||||
direct_media=no
|
||||
rtp_symmetric=yes
|
||||
force_rport=yes
|
||||
rewrite_contact=yes
|
||||
allow=alaw,h264
|
||||
transport=transport-udp
|
||||
context=internal
|
||||
aors=domofon
|
||||
auth=domofon
|
||||
|
||||
[domofon]
|
||||
type=auth
|
||||
auth_type=userpass
|
||||
password=your_password
|
||||
username=domofon
|
||||
|
||||
[domofon]
|
||||
type=aor
|
||||
max_contacts=1
|
||||
```
|
||||
|
||||
### Endpoint
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| direct_media | determines whether media may flow directly between endpoints (default: "yes")
|
||||
| rtp_symmetric | user agents (UAs) use the same socket/port binding to send and receive RTP stream packets
|
||||
| force_rport | causes Asterisk to always send responses back to the address/port from which it received requests
|
||||
| rewrite_contact | allow contact header to be rewritten with the source IP address port (default: "no")
|
||||
| allow | allow codecs in order of preference
|
||||
| transport | set the default transports; the order determines the primary default transport
|
||||
| context | default context for incoming calls (from <kbd>extensions.conf</kbd>); defaults to 'default'
|
||||
| aors | AoRs to be used with the endpoint (default: "")
|
||||
| auth | authentication objects associated with the endpoint (default: "")
|
||||
|
||||
### Auth
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| auth_type | may be "userpass" for plain text passwords or "md5" for pre-hashed credentials. (default: "userpass")
|
||||
| password, username| credentials for registration
|
||||
|
||||
|
||||
### AoR
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| max_contacts | maximum number of contacts that can bind to an AoR (default: "0")
|
||||
|
||||
## Client
|
||||
This configuration example is for registering a client application with Asterisk using WebRTC.
|
||||
|
||||
```
|
||||
[15]
|
||||
type=endpoint
|
||||
direct_media=no
|
||||
context=from-external
|
||||
allow=vp8,vp9,h264,alaw
|
||||
transport=transport-wss
|
||||
webrtc=yes
|
||||
auth=15
|
||||
aors=15
|
||||
|
||||
[15]
|
||||
type=auth
|
||||
auth_type=userpass
|
||||
password=your_password
|
||||
username=15
|
||||
|
||||
[15]
|
||||
type=aor
|
||||
max_contacts=10
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| webrtc | when set to "yes" this also enables the following values that are needed for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport
|
||||
|
||||
> VP8/VP9 are the default codecs for WebRTC and the only codec for intercom is H264.
|
||||
> In this case, a codec difference problem may occur.
|
||||
> At the time of writing only Freeswitch can transcode video from one codec to another.
|
||||
{.is-warning}
|
||||
|
||||
> You can use any name for the client.
|
||||
> I chose <kbd>15</kbd> because it is convenient to call from a smartphone with a fake client (I use Zoiper).
|
||||
{.is-info}
|
||||
|
||||
# extensions.conf
|
||||
```
|
||||
;-------------------------------------------------------
|
||||
; General Section
|
||||
;-------------------------------------------------------
|
||||
[general]
|
||||
static=yes
|
||||
writeprotect=yes
|
||||
clearglobalvars=yes
|
||||
autofallthrough=yes
|
||||
|
||||
[from-external]
|
||||
exten => 200,1,Dial(PJSIP/domofon,120)
|
||||
|
||||
[invalid]
|
||||
exten => _X,1,Stasis(hello)
|
||||
same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
same => n,Hangup
|
||||
exten => _XX,1,Stasis(hello)
|
||||
same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
same => n,Hangup
|
||||
|
||||
exten => _1[0-3]X,1,Stasis(hello)
|
||||
same => n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
same => n,Hangup
|
||||
|
||||
[default]
|
||||
```
|
||||
|
||||
`static`
|
||||
if static is set to no, or omitted, then the pbx_config will rewrite this file when extensions are modified.
|
||||
|
||||
`writeprotect`
|
||||
if static=yes and writeprotect=no, you can save dialplan by CLI command <kbd>dialplan save</kbd> too
|
||||
|
||||
`clearglobalvars`
|
||||
if clearglobalvars is set, global variables will be cleared and reparsed on a <kbd>dialplan reload</kbd>, or Asterisk reload.
|
||||
|
||||
`autofallthrough`
|
||||
if autofallthrough is set, then if an extension runs out of things to do, it will terminate the call with BUSY, CONGESTION
|
||||
if autofallthrough is not set, then if an extension runs out ofthings to do, Asterisk will wait for a new extension to be dialed
|
||||
|
||||
`_X`, `_XX`, `_1[0-3]X`
|
||||
it means that dialplans will work with 1-139 numbers
|
||||
|
||||
`${EXTEN}`
|
||||
variable equal to dialed number
|
||||
|
||||
# WebRTC
|
||||
WebRTC requires **WSS (WebSocket Secure)** and we need a domain with a certificate. You can use a self-signed certificate, but you cannot use it with a browser and sipML5.
|
||||
|
||||
## Generate certificate
|
||||
|
||||
**Let’s Encrypt** is a service offering free SSL certificates through an automated API. The most popular Let’s Encrypt client is EFF’s **Certbot**.
|
||||
|
||||
Install Certbot:
|
||||
```
|
||||
sudo add-apt-repository ppa:certbot/certbot
|
||||
sudo apt-get update
|
||||
sudo apt-get install certbot
|
||||
```
|
||||
|
||||
Certbot needs to answer a cryptographic challenge issued by the Let’s Encrypt API in order to prove we control our domain. It uses ports 80 (HTTP) or 443 (HTTPS) to accomplish this. Open up the appropriate port in your firewall:
|
||||
|
||||
```
|
||||
sudo ufw allow 80
|
||||
```
|
||||
|
||||
Substitute 443 above if that’s the port you’re using. ufw will output confirmation that your rule was added:
|
||||
|
||||
> Output
|
||||
> Rule added
|
||||
> Rule added (v6)
|
||||
{.is-success}
|
||||
|
||||
We can now run Certbot to get our certificate. We’ll use the <kbd>--standalone</kbd> option to tell Certbot to handle the challenge using its own built-in web server. The <kbd>--preferred-challenges</kbd> option instructs Certbot to use port 80 or port 443. If you’re using port 80, you want <kbd>--preferred-challenges</kbd> http. For port 443 it would be <kbd>--preferred-challenges</kbd> tls-sni. Finally, the <kbd>-d</kbd> flag is used to specify the domain you’re requesting a certificate for. You can add multiple <kbd>-d</kbd> options to cover multiple domains in one certificate.
|
||||
|
||||
```
|
||||
sudo certbot certonly --standalone --preferred-challenges http -d example.com
|
||||
```
|
||||
|
||||
## Configure Asterisk with cerificate
|
||||
|
||||
This certificate can only be accessed by the <kbd>root</kbd> user. Asterisk is executed by <kbd>asterisk</kbd> user. The way to fix this problem is to copy the certificate into Asterisk directory and change the owner.
|
||||
|
||||
If you use Asterisk as a Linux service:
|
||||
```
|
||||
mkdir /etc/asterisk/keys
|
||||
sudo cp -L /etc/letsencrypt/live/example.com/cert.pem /etc/asterisk/keys
|
||||
sudo cp -L /etc/letsencrypt/live/example.com/privkey.pem /etc/asterisk/keys
|
||||
sudo chown asterisk:asterisk /etc/asterisk/keys/cert.pem
|
||||
sudo chown asterisk:asterisk /etc/asterisk/keys/privkey.pem
|
||||
```
|
||||
|
||||
You need to enable TLS and add the certificate to <kbd>http.conf</kbd>:
|
||||
```
|
||||
[general]
|
||||
enabled=no
|
||||
tlsenable=yes
|
||||
tlsbindaddr=0.0.0.0:8089
|
||||
tlscertfile=/etc/asterisk/keys/cert.pem
|
||||
tlsprivatekey=/etc/asterisk/keys/privkey.pem
|
||||
```
|
||||
|
||||
## Testing
|
||||
|
||||
You can use one of 2 sites to test WebRTC from your browser:
|
||||
- https://www.doubango.org/sipml5/call.htm
|
||||
- https://tryit.jssip.net/
|
||||
|
||||
# HTTP server (ARI)
|
||||
The HTTP server in Asterisk is configured via http.conf. Note that this does not describe all of the options available via http.conf - rather, it lists the most useful ones for ARI.
|
||||
|
||||
Example:
|
||||
```
|
||||
[general]
|
||||
enabled = yes
|
||||
bindaddr = 0.0.0.0
|
||||
bindport = 8088
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| enabled | Enable the HTTP server. The HTTP server in Asterisk is disabled by default. Unless it is enabled, ARI will not function!
|
||||
| bindaddr | The IP address to bind the HTTP server to. This can either be an explicit local address, or 0.0.0.0 to bind to all available interfaces.
|
||||
| bindport | The port to bind the HTTP server to. Client making HTTP requests should specify 8088 as the port to send the request to.
|
||||
| prefix | A prefix to require for all requests. If specified, requests must begin with the specified prefix.
|
||||
| tlsenable | Enable HTTPS
|
||||
| tlsbindaddr | The IP address and port to bind the HTTPS server to. This should be an IP address and port, e.g., 0.0.0.0:8089
|
||||
| tlscertfile | The full path to the certificate file to use. Asterisk only supports the .pem format
|
||||
| tlsprivatekey| The full path to the private key file. Asterisk only supports the .pem format. If this is not specified, the certificate specified in tlscertfile will be searched for the private key.
|
||||
|
||||
For creating a certificate, see [WebRTC configuration](/telecom/Asterisk/webrtc)
|
||||
|
||||
# Fail2Ban
|
||||
|
||||
To install fail2ban:
|
||||
```
|
||||
sudo apt install fail2ban
|
||||
```
|
||||
|
||||
The default settings of the program are in the <kbd>/etc/fail2ban/jail.conf</kbd> file, it is recommended to change the settings in <kbd>/etc/fail2ban/jail.local</kbd>, which is a copy of <kbd>jail.conf</kbd>.
|
||||
|
||||
The file contains a section of general settings <kbd>[DEFAULT]</kbd> and sections of specific settings for certain services (for example, the presence of the <kbd>[ssh]</kbd> section is demonstrated).
|
||||
|
||||
```
|
||||
[DEFAULT]
|
||||
ignoreip = 127.0.0.1/8
|
||||
ignorecommand =
|
||||
bantime = 3600
|
||||
findtime = 600
|
||||
maxretry = 3
|
||||
backend = auto
|
||||
usedns = warn
|
||||
destemail = root@localhost
|
||||
sendername = Fail2Ban
|
||||
sender = fail2ban@localhost
|
||||
banaction = iptables-multiport
|
||||
mta = sendmail
|
||||
protocol = tcp
|
||||
chain = INPUT
|
||||
action_ = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
action_mw = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
%(mta)s-whois[name=%(__name__)s, dest="%(destemail)s", protocol="%(protocol)s", chain="%(chain)s", sendername="%(sendername)s"]
|
||||
action_mwl = %(banaction)s[name=%(__name__)s, port="%(port)s", protocol="%(protocol)s", chain="%(chain)s"]
|
||||
%(mta)s-whois-lines[name=%(__name__)s, dest="%(destemail)s", logpath=%(logpath)s, chain="%(chain)s", sendername="%(sendername)s"]
|
||||
action = %(action_)s
|
||||
|
||||
[ssh]
|
||||
enabled = true
|
||||
port = ssh
|
||||
filter = sshd
|
||||
logpath = /var/log/auth.log
|
||||
maxretry = 6
|
||||
```
|
||||
|
||||
| Variable | Description |
|
||||
| ----------- | ----------- |
|
||||
| ignoreip | can be a list of IP addresses, CIDR masks or DNS hosts. Fail2ban will not ban a host which matches an address in this list. Several addresses can be defined using space (and/or comma) separator.
|
||||
| bantime | the number of seconds that a host is banned.
|
||||
| findtime, maxretry| a host is banned if it has generated "maxretry" during the last "findtime" seconds
|
||||
| backend | specifies the backend used to get files modification
|
||||
| usedns | specifies if jails should trust hostnames in logs, warn when DNS lookups are performed, or ignore all hostnames in logsyes: if a hostname is encountered, a DNS lookup will be performed
|
||||
| destemail | destination email address used solely for the interpolations in jail.{conf,local,d/*} configuration files.
|
||||
| sender | sender email address used solely for some actions
|
||||
| banaction | default banning action (e.g. iptables, iptables-new, iptables-multiport, shorewall, etc) It is used to define a ction_* variables
|
||||
| protocol | default protocol (tcp, udp, ...)
|
||||
| chain | specify chain where jumps would need to be added in ban-actions expecting parameter chain
|
||||
| enabled | enables the jails
|
||||
| port | ports to be banned
|
||||
| filter | defines the filter to use by the jail
|
||||
| logpath | path to the logs
|
||||
|
||||
## Asterisk configuration
|
||||
|
||||
1. Copy the configuration file to the local one.
|
||||
```
|
||||
sudo cp /etc/fail2ban/jail.conf /etc/fail2ban/jail.local
|
||||
```
|
||||
|
||||
2. Add to <kbd>ignoreip</kbd> the addresses from which you are going to connect to the asterisk.
|
||||
|
||||
3. Add Asterisk jail (see below). You can set up variables however you like.
|
||||
|
||||
```
|
||||
[asterisk]
|
||||
enabled = true
|
||||
filter = asterisk
|
||||
backend = auto
|
||||
port = 5060,5061
|
||||
action = iptables-allports[name=ASTERISK, protocol=all, blocktype=DROP]
|
||||
logpath = /var/log/asterisk/messages
|
||||
findtime = 1m
|
||||
maxretry = 5
|
||||
bantime = 30d
|
||||
```
|
||||
|
||||
4. (optional) If you have ssh, it is more secure if you add it to the jail.
|
||||
|
||||
```
|
||||
[sshd]
|
||||
enabled = true
|
||||
bantime = 60m
|
||||
findtime = 1m
|
||||
maxretry = 5
|
||||
port = ssh
|
||||
logpath = %(sshd_log)s
|
||||
backend = %(sshd_backend)s
|
||||
```
|
||||
80
asterisk/acl.conf
Executable file
80
asterisk/acl.conf
Executable file
@ -0,0 +1,80 @@
|
||||
;
|
||||
; Named Access Control Lists (ACLs)
|
||||
;
|
||||
; A convenient way to share acl definitions
|
||||
;
|
||||
; This configuration file is read on startup
|
||||
;
|
||||
; CLI Commands
|
||||
; -----------------------------------------------------------
|
||||
; acl show Show all named ACLs configured
|
||||
; acl show <name> Show contents of a particular named ACL
|
||||
; reload acl Reload configuration file
|
||||
;
|
||||
; Any configuration that uses ACLs which has been made to be able to use named
|
||||
; ACLs will specify a named ACL with the 'acl' option in its configuration in
|
||||
; a similar fashion to the usual 'permit' and 'deny' options. Example:
|
||||
; acl=my_named_acl
|
||||
;
|
||||
; Multiple named ACLs can be applied by either comma separating the arguments or
|
||||
; just by adding additional ACL lines. Example:
|
||||
; acl=my_named_acl
|
||||
; acl=my_named_acl2
|
||||
;
|
||||
; or
|
||||
;
|
||||
; acl=my_named_acl,my_named_acl2
|
||||
;
|
||||
; ACLs specified by name are evaluated independently from the ACL specified via
|
||||
; permit/deny. In order for an address to pass a given ACL, it must pass both
|
||||
; the ACL specified by permit/deny for a given item as well as any named ACLs
|
||||
; that were specified.
|
||||
;
|
||||
;[example_named_acl1]
|
||||
;deny=0.0.0.0/0.0.0.0
|
||||
;permit=209.16.236.0
|
||||
;permit=209.16.236.1
|
||||
;
|
||||
;[example_named_acl2]
|
||||
;permit=0.0.0.0/0.0.0.0
|
||||
;deny=10.24.20.171
|
||||
;deny=10.24.20.103
|
||||
;deny=209.16.236.1
|
||||
;
|
||||
; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
|
||||
; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
|
||||
; The following example explains how combining the ACLs works:
|
||||
; <in another configuration>
|
||||
; [example_item_with_acl]
|
||||
; acl=example_named_acl1
|
||||
; acl=example_named_acl2
|
||||
;
|
||||
; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
|
||||
; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
|
||||
; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
|
||||
; either, so it passes.
|
||||
;
|
||||
; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
|
||||
; First, example_named_acl1 is evaluated and the address is allowed.
|
||||
; However, it is blocked by example_named_acl2, so the address is blocked from the combined
|
||||
; ACL.
|
||||
;
|
||||
; Similarly, the permits/denies in specific configurations that make up an ACL definition
|
||||
; are also treated as a separate ACL for evaluation. So if we change the example above to:
|
||||
; <in another configuration>
|
||||
; [example_item_with_acl]
|
||||
; acl=example_named_acl1
|
||||
; acl=example_named_acl2
|
||||
; deny=209.16.236.0
|
||||
;
|
||||
; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
|
||||
; though it passes the two named components.
|
||||
;
|
||||
;
|
||||
; Named ACLs can use ipv6 addresses just like normal ACLs.
|
||||
;[ipv6_example_1]
|
||||
;deny = ::
|
||||
;permit = ::1/128
|
||||
;
|
||||
;[ipv6_example_2]
|
||||
;permit = fe80::21d:bad:fad:2323
|
||||
8
asterisk/adsi.conf
Executable file
8
asterisk/adsi.conf
Executable file
@ -0,0 +1,8 @@
|
||||
;
|
||||
; Sample ADSI Configuration file
|
||||
;
|
||||
[intro]
|
||||
alignment = center
|
||||
greeting => Welcome to the
|
||||
greeting => Asterisk
|
||||
greeting => Open Source PBX
|
||||
70
asterisk/agents.conf
Executable file
70
asterisk/agents.conf
Executable file
@ -0,0 +1,70 @@
|
||||
;
|
||||
; Agent pool configuration
|
||||
;
|
||||
|
||||
[general]
|
||||
; The general section of this config is not currently used, but reserved
|
||||
; for future use.
|
||||
|
||||
;[agent-id]
|
||||
; Define ackcall to require the agent to give a DTMF acknowledgement
|
||||
; when the agent receives a call.
|
||||
; The channel variable AGENTACKCALL overrides on agent login.
|
||||
; Default is "no".
|
||||
;ackcall=no
|
||||
;
|
||||
; Set what DTMF key sequence the agent should use to acknowledge a call.
|
||||
; The channel variable AGENTACCEPTDTMF overrides on agent login.
|
||||
; This option is ignored unless ackcall is enabled.
|
||||
; Default is "#".
|
||||
;acceptdtmf=##
|
||||
;
|
||||
; Set how many seconds a call for the agent has to wait for the agent to
|
||||
; acknowledge the call before the agent is automatically logged off. If
|
||||
; set to zero then the call will wait forever for the agent to acknowledge.
|
||||
; The channel variable AGENTAUTOLOGOFF overrides on agent login.
|
||||
; This option is ignored unless ackcall is enabled.
|
||||
; Default is 0.
|
||||
;autologoff=15
|
||||
;
|
||||
; Set the minimum amount of time after disconnecting a call before
|
||||
; the agent can receive a new call in milliseconds.
|
||||
; The channel variable AGENTWRAPUPTIME overrides on agent login.
|
||||
; Default is 0.
|
||||
;wrapuptime=5000
|
||||
;
|
||||
; Set the musiconhold class for the agent.
|
||||
; Default is "default".
|
||||
;musiconhold=default
|
||||
;
|
||||
; Enable recording calls the agent takes automatically by invoking the
|
||||
; DTMF automixmon feature when the agent connects to a caller.
|
||||
; See features.conf.sample for information about the automixmon feature.
|
||||
; Default is "no".
|
||||
;recordagentcalls=yes
|
||||
;
|
||||
; The sound file played to alert the agent when a call is present.
|
||||
; Default is "beep".
|
||||
;custom_beep=beep
|
||||
;
|
||||
; A friendly name for the agent used in log messages.
|
||||
; Default is "".
|
||||
;fullname=Mark Spencer
|
||||
;
|
||||
; --------------------------------------------------
|
||||
;
|
||||
; This section contains example agent definitions:
|
||||
;
|
||||
; Define a template called my-agents:
|
||||
;[my-agents](!)
|
||||
;autologoff=15
|
||||
;ackcall=yes
|
||||
;acceptdtmf=##
|
||||
;
|
||||
; Define agent 1001 using the my-agents template:
|
||||
;[1001](my-agents)
|
||||
;fullname=Mark Spencer
|
||||
;
|
||||
; Define agent 1002 using the my-agents template:
|
||||
;[1002](my-agents)
|
||||
;fullname=Will Meadows
|
||||
91
asterisk/alarmreceiver.conf
Executable file
91
asterisk/alarmreceiver.conf
Executable file
@ -0,0 +1,91 @@
|
||||
;
|
||||
; alarmreceiver.conf
|
||||
;
|
||||
; Sample configuration file for the Asterisk alarm receiver application.
|
||||
;
|
||||
|
||||
|
||||
[general]
|
||||
|
||||
;
|
||||
; Specify a timestamp format for the metadata section of the event files
|
||||
; Default is %a %b %d, %Y @ %H:%M:%S %Z
|
||||
|
||||
timestampformat = %a %b %d, %Y @ %H:%M:%S %Z
|
||||
|
||||
;
|
||||
; Specify a command to execute when the caller hangs up
|
||||
;
|
||||
; Default is none
|
||||
;
|
||||
|
||||
;eventcmd = yourprogram -yourargs ...
|
||||
|
||||
;
|
||||
; Specify a spool directory for the event files. This setting is required
|
||||
; if you want the app to be useful. Event files written to the spool
|
||||
; directory will be of the template event-XXXXXX, where XXXXXX is a random
|
||||
; and unique alphanumeric string.
|
||||
;
|
||||
; Default is none, and the events will be dropped on the floor.
|
||||
;
|
||||
|
||||
eventspooldir = /tmp
|
||||
|
||||
;
|
||||
; The alarmreceiver app can either log the events one-at-a-time to individual
|
||||
; files in the spool directory, or it can store them until the caller
|
||||
; disconnects and write them all to one file.
|
||||
;
|
||||
; The default setting for logindividualevents is no.
|
||||
;
|
||||
|
||||
logindividualevents = no
|
||||
|
||||
;
|
||||
; The timeout for receiving the first DTMF digit is adjustable from 1000 msec.
|
||||
; to 10000 msec. The default is 2000 msec. Note: if you wish to test the
|
||||
; receiver by entering digits manually, set this to a reasonable time out
|
||||
; like 10000 milliseconds.
|
||||
|
||||
fdtimeout = 2000
|
||||
|
||||
;
|
||||
; The timeout for receiving subsequent DTMF digits is adjustable from
|
||||
; 110 msec. to 4000 msec. The default is 200 msec. Note: if you wish to test
|
||||
; the receiver by entering digits manually, set this to a reasonable time out
|
||||
; like 4000 milliseconds.
|
||||
;
|
||||
|
||||
sdtimeout = 200
|
||||
|
||||
;
|
||||
; Wait for the connection to settle post-answer. Adjustable from 500 msec. to 10000 msec.
|
||||
; The default is 1250 msec.
|
||||
;
|
||||
|
||||
answait = 1250
|
||||
|
||||
; When logging individual events it may be desirable to skip grouping of metadata
|
||||
|
||||
;no_group_meta = yes
|
||||
|
||||
;
|
||||
; The loudness of the ACK and Kissoff tones is adjustable from 100 to 8192.
|
||||
; The default is 8192. This shouldn't need to be messed with, but is included
|
||||
; just in case there are problems with signal levels.
|
||||
;
|
||||
|
||||
loudness = 8192
|
||||
|
||||
;
|
||||
; The db-family setting allows the user to capture statistics on the number of
|
||||
; calls, and the errors the alarm receiver sees. The default is for no
|
||||
; db-family name to be defined and the database logging to be turned off.
|
||||
;
|
||||
|
||||
;db-family = yourfamily:
|
||||
|
||||
;
|
||||
; End of alarmreceiver.conf
|
||||
;
|
||||
77
asterisk/alsa.conf
Executable file
77
asterisk/alsa.conf
Executable file
@ -0,0 +1,77 @@
|
||||
;
|
||||
; Open Sound System Console Driver Configuration File
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; Automatically answer incoming calls on the console? Choose yes if
|
||||
; for example you want to use this as an intercom.
|
||||
;
|
||||
autoanswer=yes
|
||||
;
|
||||
; Default context (is overridden with @context syntax)
|
||||
;
|
||||
context=local
|
||||
;
|
||||
; Default extension to call
|
||||
;
|
||||
extension=s
|
||||
;
|
||||
; Default language
|
||||
;
|
||||
;language=en
|
||||
;
|
||||
; Default Music on Hold class to use when this channel is placed on hold in
|
||||
; the case that the music class is not set on the channel with
|
||||
; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
|
||||
; putting this one on hold did not suggest a class to use.
|
||||
;
|
||||
;mohinterpret=default
|
||||
;
|
||||
; Silence suppression can be enabled when sound is over a certain threshold.
|
||||
; The value for the threshold should probably be between 500 and 2000 or so,
|
||||
; but your mileage may vary. Use the echo test to evaluate the best setting.
|
||||
;silencesuppression = yes
|
||||
;silencethreshold = 1000
|
||||
;
|
||||
; To set which ALSA device to use, change this parameter
|
||||
;input_device=hw:0,0
|
||||
;output_device=hw:0,0
|
||||
|
||||
;
|
||||
; Default mute state (can also be toggled via CLI)
|
||||
;mute=true
|
||||
|
||||
;
|
||||
; If enabled, no audio capture device will be opened. This is useful on
|
||||
; systems where there will be no return audio path, such as overhead pagers.
|
||||
;noaudiocapture=true
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; ALSA channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The ALSA channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive ALSA side will always
|
||||
; be used if the sending side can create jitter.
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
27
asterisk/amd.conf
Executable file
27
asterisk/amd.conf
Executable file
@ -0,0 +1,27 @@
|
||||
;
|
||||
; Answering Machine Detection Configuration
|
||||
;
|
||||
|
||||
[general]
|
||||
total_analysis_time = 5000 ; Maximum time allowed for the algorithm to decide
|
||||
; on whether the audio represents a HUMAN, or a MACHINE
|
||||
silence_threshold = 256 ; If the average level of noise in a sample does not reach
|
||||
; this value, from a scale of 0 to 32767, then we will consider
|
||||
; it to be silence.
|
||||
|
||||
; Greeting ;
|
||||
initial_silence = 2500 ; Maximum silence duration before the greeting.
|
||||
; If exceeded, then the result is detection as a MACHINE.
|
||||
after_greeting_silence = 800 ; Silence after detecting a greeting.
|
||||
; If exceeded, then the result is detection as a HUMAN
|
||||
greeting = 1500 ; Maximum length of a greeting. If exceeded, then the
|
||||
; result is detection as a MACHINE.
|
||||
|
||||
; Word detection ;
|
||||
min_word_length = 100 ; Minimum duration of Voice to considered as a word
|
||||
maximum_word_length = 5000 ; Maximum duration of a single Voice utterance allowed.
|
||||
between_words_silence = 50 ; Minimum duration of silence after a word to consider
|
||||
; the audio what follows as a new word
|
||||
|
||||
maximum_number_of_words = 2 ; Maximum number of words in the greeting
|
||||
; If exceeded, then the result is detection as a MACHINE
|
||||
24
asterisk/app_mysql.conf
Executable file
24
asterisk/app_mysql.conf
Executable file
@ -0,0 +1,24 @@
|
||||
; Configuration file for the MYSQL app addon
|
||||
|
||||
[general]
|
||||
;
|
||||
; Nullvalue governs how NULL values are returned from the database. In
|
||||
; previous versions, the special NULL value was returned as the "NULL"
|
||||
; string. We now provide an option for the behavior, configured globally.
|
||||
; nullstring - the string "NULL"
|
||||
; emptystring - the string ""
|
||||
; null - unset the variable
|
||||
;
|
||||
; WARNING: setting nullvalue=null may have undesireable consequences, in
|
||||
; particular if you use subroutines in AEL or the LOCAL() variable construct.
|
||||
; You have been warned. Don't complain if you use that setting in combination
|
||||
; with Gosub or AEL and get buggy behavior.
|
||||
;
|
||||
nullvalue = nullstring
|
||||
|
||||
; If set, autoclear will destroy allocated statement and connection resources
|
||||
; when the channel ends. For most usage of the MYSQL app, this is what you
|
||||
; want, but it's conceivable that somebody is sharing MYSQL connections across
|
||||
; multiple channels, in which case, this should be set to 'no'. Defaults to
|
||||
; 'no', as this was the original behavior.
|
||||
autoclear=yes
|
||||
27
asterisk/app_skel.conf
Executable file
27
asterisk/app_skel.conf
Executable file
@ -0,0 +1,27 @@
|
||||
[general]
|
||||
games=3
|
||||
cheat=no
|
||||
|
||||
[sounds]
|
||||
prompt=please-enter-your,number,queue-less-than
|
||||
wrong_guess=vm-pls-try-again
|
||||
right_guess=auth-thankyou
|
||||
too_high=high
|
||||
too_low=low
|
||||
lose=vm-goodbye
|
||||
|
||||
[easy]
|
||||
max_number=10
|
||||
max_guesses=4
|
||||
|
||||
[medium]
|
||||
max_number=100
|
||||
max_guesses=6
|
||||
|
||||
[hard]
|
||||
max_number=1000
|
||||
max_guesses=7
|
||||
|
||||
[nightmare]
|
||||
max_number=1000
|
||||
max_guesses=1
|
||||
15
asterisk/ari.conf
Executable file
15
asterisk/ari.conf
Executable file
@ -0,0 +1,15 @@
|
||||
[general]
|
||||
enabled=yes
|
||||
;pretty=yes ; we don't need pretty-printing of the JSON responses in this
|
||||
; example, but you might if you use curl a lot.
|
||||
;
|
||||
; In this example, we are going to use the version of Swagger-UI that is hosted
|
||||
; at ari.asterisk.org. In order to get past CORS restrictions in the browser,
|
||||
; That origin needs to be added to the allowed_origins list.
|
||||
;
|
||||
allowed_origins=http://ari.asterisk.org
|
||||
|
||||
[hey]
|
||||
type=user
|
||||
password=jvehDRCWRwWQCGdF
|
||||
;read_only=no ; Set to yes for read-only applications
|
||||
37
asterisk/ari.conf.default
Executable file
37
asterisk/ari.conf.default
Executable file
@ -0,0 +1,37 @@
|
||||
[general]
|
||||
enabled = yes ; When set to no, ARI support is disabled.
|
||||
;pretty = no ; When set to yes, responses from ARI are
|
||||
; ; formatted to be human readable.
|
||||
;allowed_origins = ; Comma separated list of allowed origins, for
|
||||
; ; Cross-Origin Resource Sharing. May be set to * to
|
||||
; ; allow all origins.
|
||||
;auth_realm = ; Realm to use for authentication. Defaults to Asterisk
|
||||
; ; REST Interface.
|
||||
;
|
||||
; Default write timeout to set on websockets. This value may need to be adjusted
|
||||
; for connections where Asterisk must write a substantial amount of data and the
|
||||
; receiving clients are slow to process the received information. Value is in
|
||||
; milliseconds; default is 100 ms.
|
||||
;websocket_write_timeout = 100
|
||||
;
|
||||
; Display certain channel variables every time a channel-oriented
|
||||
; event is emitted:
|
||||
;
|
||||
; Note that this does incur a performance penalty and should be avoided if possible.
|
||||
;
|
||||
;channelvars = var1,var2,var3
|
||||
|
||||
;[username]
|
||||
;type = user ; Specifies user configuration
|
||||
;read_only = no ; When set to yes, user is only authorized for
|
||||
; ; read-only requests.
|
||||
;
|
||||
;password = ; Crypted or plaintext password (see password_format).
|
||||
;
|
||||
; password_format may be set to plain (the default) or crypt. When set to crypt,
|
||||
; crypt(3) is used to validate the password. A crypted password can be generated
|
||||
; using mkpasswd -m sha-512.
|
||||
;
|
||||
; When set to plain, the password is in plaintext.
|
||||
;
|
||||
;password_format = plain
|
||||
63
asterisk/ast_debug_tools.conf
Executable file
63
asterisk/ast_debug_tools.conf
Executable file
@ -0,0 +1,63 @@
|
||||
#
|
||||
# This file is used by the Asterisk debug tools.
|
||||
# Unlike other Asterisk config files, this one is
|
||||
# "sourced" by bash and must adhere to bash semantics.
|
||||
#
|
||||
|
||||
# A list of coredumps and/or coredump search patterns.
|
||||
# Bash extended globs are enabled and any resulting files
|
||||
# that aren't actually coredumps are silently ignored
|
||||
# so you can be liberal with the globs.
|
||||
#
|
||||
# If your patterns contains spaces be sure to only quote
|
||||
# the portion of the pattern that DOESN'T contain wildcard
|
||||
# expressions. If you quote the whole pattern, it won't
|
||||
# be expanded and the glob characters will be treated as
|
||||
# literals.
|
||||
#
|
||||
# The exclusion of files ending ".txt" is just for
|
||||
# demonstration purposes as non-coredumps will be ignored
|
||||
# anyway.
|
||||
COREDUMPS=(/tmp/core[-._]asterisk!(*.txt) /tmp/core[-._]$(hostname)!(*.txt))
|
||||
|
||||
# The directory to contain output files and work directories.
|
||||
# For output from existing core files, the default is the
|
||||
# directory that the core file is found in. For core files
|
||||
# produced from a running process, the default is /tmp.
|
||||
OUTPUTDIR=/tmp
|
||||
|
||||
# Date command for the "running" coredump and tarballs.
|
||||
# DATEFORMAT will be executed to get the timestamp.
|
||||
# Don't put quotes around the format string or they'll be
|
||||
# treated as literal characters. Also be aware of colons
|
||||
# in the output as you can't upload files with colons in
|
||||
# the name to Jira.
|
||||
#
|
||||
# Unix timestamp
|
||||
#DATEFORMAT='date +%s.%N'
|
||||
#
|
||||
# Unix timestamp on *BSD/MacOS after installing coreutils
|
||||
#DATEFORMAT='gdate +%s.%N'
|
||||
#
|
||||
# Readable GMT
|
||||
#DATEFORMAT='date -u +%FT%H-%M-%S%z'
|
||||
#
|
||||
# Readable Local time
|
||||
DATEFORMAT='date +%FT%H-%M-%S%z'
|
||||
|
||||
# A list of log files and/or log file search patterns using the
|
||||
# same syntax as COREDUMPS.
|
||||
#
|
||||
LOGFILES=(/var/log/asterisk/messages* /var/log/asterisk/queue* \
|
||||
/var/log/asterisk/debug* /var/log/asterisk/security*)
|
||||
|
||||
# ast_loggrabber converts POSIX timestamps to readable format
|
||||
# using this Python strftime format string. If not specified
|
||||
# or an empty string, no format covnersion is done.
|
||||
LOG_DATEFORMAT="%m/%d-%H:%M:%S.%f"
|
||||
|
||||
# The timezone to use when converting POSIX timestamps to
|
||||
# readable format. It can be specified in "<continent>/<city>"
|
||||
# format or in abbreviation format such as "CST6CDT". If not
|
||||
# specified, the "local" timezone is used.
|
||||
# LOG_TIMEZONE=
|
||||
158
asterisk/asterisk.adsi
Executable file
158
asterisk/asterisk.adsi
Executable file
@ -0,0 +1,158 @@
|
||||
;
|
||||
; Asterisk default ADSI script
|
||||
;
|
||||
;
|
||||
; Begin with the preamble requirements
|
||||
;
|
||||
DESCRIPTION "Asterisk PBX" ; Name of vendor
|
||||
VERSION 0x00 ; Version of stuff
|
||||
;SECURITY "_AST" ; Security code
|
||||
SECURITY 0X9BDBF7AC ; Security code
|
||||
FDN 0x0000000F ; Descriptor number
|
||||
|
||||
;
|
||||
; Flags
|
||||
;
|
||||
FLAG "nocallwaiting"
|
||||
|
||||
;
|
||||
; Predefined strings
|
||||
;
|
||||
DISPLAY "titles" IS "** Asterisk PBX **"
|
||||
DISPLAY "talkingto" IS "Call active." JUSTIFY LEFT
|
||||
DISPLAY "callname" IS "$Call1p" JUSTIFY LEFT
|
||||
DISPLAY "callnum" IS "$Call1s" JUSTIFY LEFT
|
||||
DISPLAY "incoming" IS "Incoming call!" JUSTIFY LEFT
|
||||
DISPLAY "ringing" IS "Calling... " JUSTIFY LEFT
|
||||
DISPLAY "callended" IS "Call ended." JUSTIFY LEFT
|
||||
DISPLAY "missedcall" IS "Missed call." JUSTIFY LEFT
|
||||
DISPLAY "busy" IS "Busy." JUSTIFY LEFT
|
||||
DISPLAY "reorder" IS "Reorder." JUSTIFY LEFT
|
||||
DISPLAY "cwdisabled" IS "Callwait disabled"
|
||||
DISPLAY "empty" IS "asdf"
|
||||
|
||||
;
|
||||
; Begin soft key definitions
|
||||
;
|
||||
KEY "callfwd" IS "CallFwd" OR "Call Forward"
|
||||
OFFHOOK
|
||||
VOICEMODE
|
||||
WAITDIALTONE
|
||||
SENDDTMF "*60"
|
||||
GOTO "offHook"
|
||||
ENDKEY
|
||||
|
||||
KEY "vmail_OH" IS "VMail" OR "Voicemail"
|
||||
OFFHOOK
|
||||
VOICEMODE
|
||||
WAITDIALTONE
|
||||
SENDDTMF "8500"
|
||||
ENDKEY
|
||||
|
||||
KEY "vmail" IS "VMail" OR "Voicemail"
|
||||
SENDDTMF "8500"
|
||||
ENDKEY
|
||||
|
||||
KEY "backspace" IS "BackSpc" OR "Backspace"
|
||||
BACKSPACE
|
||||
ENDKEY
|
||||
|
||||
KEY "cwdisable" IS "CWDsble" OR "Disable Call Wait"
|
||||
SENDDTMF "*70"
|
||||
SETFLAG "nocallwaiting"
|
||||
SHOWDISPLAY "cwdisabled" AT 4
|
||||
TIMERCLEAR
|
||||
TIMERSTART 1
|
||||
ENDKEY
|
||||
|
||||
KEY "cidblock" IS "CIDBlk" OR "Block Callerid"
|
||||
SENDDTMF "*67"
|
||||
SETFLAG "nocallwaiting"
|
||||
ENDKEY
|
||||
|
||||
;
|
||||
; Begin main subroutine
|
||||
;
|
||||
|
||||
SUB "main" IS
|
||||
IFEVENT NEARANSWER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "talkingto" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" AT 3
|
||||
SHOWDISPLAY "callnum" AT 4
|
||||
GOTO "stableCall"
|
||||
ENDIF
|
||||
IFEVENT OFFHOOK THEN
|
||||
CLEAR
|
||||
CLEARFLAG "nocallwaiting"
|
||||
CLEARDISPLAY
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail"
|
||||
SHOWKEYS "cidblock"
|
||||
SHOWKEYS "cwdisable" UNLESS "nocallwaiting"
|
||||
GOTO "offHook"
|
||||
ENDIF
|
||||
IFEVENT IDLE THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail_OH"
|
||||
ENDIF
|
||||
IFEVENT CALLERID THEN
|
||||
CLEAR
|
||||
; SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
; SHOWDISPLAY "incoming" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" AT 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" AT 4
|
||||
ENDIF
|
||||
IFEVENT RING THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "incoming" AT 2
|
||||
ENDIF
|
||||
IFEVENT ENDOFRING THEN
|
||||
SHOWDISPLAY "missedcall" AT 2
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1
|
||||
SHOWKEYS "vmail_OH"
|
||||
ENDIF
|
||||
IFEVENT TIMER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "empty" AT 4
|
||||
ENDIF
|
||||
ENDSUB
|
||||
|
||||
SUB "offHook" IS
|
||||
IFEVENT FARRING THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "ringing" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
IFEVENT FARANSWER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "talkingto" AT 2
|
||||
GOTO "stableCall"
|
||||
ENDIF
|
||||
IFEVENT BUSY THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "busy" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
IFEVENT REORDER THEN
|
||||
CLEAR
|
||||
SHOWDISPLAY "titles" AT 1 NOUPDATE
|
||||
SHOWDISPLAY "reorder" AT 2 NOUPDATE
|
||||
SHOWDISPLAY "callname" at 3 NOUPDATE
|
||||
SHOWDISPLAY "callnum" at 4
|
||||
ENDIF
|
||||
ENDSUB
|
||||
|
||||
SUB "stableCall" IS
|
||||
IFEVENT REORDER THEN
|
||||
SHOWDISPLAY "callended" AT 2
|
||||
ENDIF
|
||||
ENDSUB
|
||||
133
asterisk/asterisk.conf
Executable file
133
asterisk/asterisk.conf
Executable file
@ -0,0 +1,133 @@
|
||||
[directories](!)
|
||||
astetcdir => /etc/asterisk
|
||||
astmoddir => /usr/lib/asterisk/modules
|
||||
astvarlibdir => /var/lib/asterisk
|
||||
astdbdir => /var/lib/asterisk
|
||||
astkeydir => /var/lib/asterisk
|
||||
astdatadir => /var/lib/asterisk
|
||||
astagidir => /var/lib/asterisk/agi-bin
|
||||
astspooldir => /var/spool/asterisk
|
||||
astrundir => /var/run/asterisk
|
||||
astlogdir => /var/log/asterisk
|
||||
astsbindir => /usr/sbin
|
||||
|
||||
[options]
|
||||
verbose = 3
|
||||
debug = 3
|
||||
;trace = 0 ; Set the trace level.
|
||||
;refdebug = yes ; Enable reference count debug logging.
|
||||
;alwaysfork = yes ; Same as -F at startup.
|
||||
;nofork = yes ; Same as -f at startup.
|
||||
;quiet = yes ; Same as -q at startup.
|
||||
;timestamp = yes ; Same as -T at startup.
|
||||
;execincludes = yes ; Support #exec in config files.
|
||||
;console = yes ; Run as console (same as -c at startup).
|
||||
;highpriority = yes ; Run realtime priority (same as -p at
|
||||
; startup).
|
||||
;initcrypto = yes ; Initialize crypto keys (same as -i at
|
||||
; startup).
|
||||
;nocolor = yes ; Disable console colors.
|
||||
;dontwarn = yes ; Disable some warnings.
|
||||
;dumpcore = yes ; Dump core on crash (same as -g at startup).
|
||||
;languageprefix = yes ; Use the new sound prefix path syntax.
|
||||
;systemname = my_system_name ; Prefix uniqueid with a system name for
|
||||
; Global uniqueness issues.
|
||||
;autosystemname = yes ; Automatically set systemname to hostname,
|
||||
; uses 'localhost' on failure, or systemname if
|
||||
; set.
|
||||
;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
|
||||
; If we get shorter DTMF messages, these will be
|
||||
; changed to the minimum duration
|
||||
;maxcalls = 10 ; Maximum amount of calls allowed.
|
||||
;maxload = 0.9 ; Asterisk stops accepting new calls if the
|
||||
; load average exceed this limit.
|
||||
;maxfiles = 1000 ; Maximum amount of openfiles.
|
||||
;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if
|
||||
; the amount of free memory falls below this
|
||||
; watermark.
|
||||
;cache_media_frames = yes ; Cache media frames for performance
|
||||
; Disable this option to help track down media frame
|
||||
; mismanagement when using valgrind or MALLOC_DEBUG.
|
||||
; The cache gets in the way of determining if the
|
||||
; frame is used after being freed and who freed it.
|
||||
; NOTE: This option has no effect when Asterisk is
|
||||
; compiled with the LOW_MEMORY compile time option
|
||||
; enabled because the cache code does not exist.
|
||||
; Default yes
|
||||
;cache_record_files = yes ; Cache recorded sound files to another
|
||||
; directory during recording.
|
||||
;record_cache_dir = /tmp ; Specify cache directory (used in conjunction
|
||||
; with cache_record_files).
|
||||
;transmit_silence = yes ; Transmit silence while a channel is in a
|
||||
; waiting state, a recording only state, or
|
||||
; when DTMF is being generated. Note that the
|
||||
; silence internally is generated in raw signed
|
||||
; linear format. This means that it must be
|
||||
; transcoded into the native format of the
|
||||
; channel before it can be sent to the device.
|
||||
; It is for this reason that this is optional,
|
||||
; as it may result in requiring a temporary
|
||||
; codec translation path for a channel that may
|
||||
; not otherwise require one.
|
||||
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
|
||||
; directly.
|
||||
;runuser = asterisk ; The user to run as.
|
||||
;rungroup = asterisk ; The group to run as.
|
||||
;lightbackground = yes ; If your terminal is set for a light-colored
|
||||
; background.
|
||||
;forceblackbackground = yes ; Force the background of the terminal to be
|
||||
; black, in order for terminal colors to show
|
||||
; up properly.
|
||||
;defaultlanguage = en ; Default language
|
||||
documentation_language = en_US ; Set the language you want documentation
|
||||
; displayed in. Value is in the same format as
|
||||
; locale names.
|
||||
;hideconnect = yes ; Hide messages displayed when a remote console
|
||||
; connects and disconnects.
|
||||
;lockconfdir = no ; Protect the directory containing the
|
||||
; configuration files (/etc/asterisk) with a
|
||||
; lock.
|
||||
;stdexten = gosub ; How to invoke the extensions.conf stdexten.
|
||||
; macro - Invoke the stdexten using a macro as
|
||||
; done by legacy Asterisk versions.
|
||||
; gosub - Invoke the stdexten using a gosub as
|
||||
; documented in extensions.conf.sample.
|
||||
; Default gosub.
|
||||
;live_dangerously = no ; Enable the execution of 'dangerous' dialplan
|
||||
; functions from external sources (AMI,
|
||||
; etc.) These functions (such as SHELL) are
|
||||
; considered dangerous because they can allow
|
||||
; privilege escalation.
|
||||
; Default no
|
||||
;entityid=00:11:22:33:44:55 ; Entity ID.
|
||||
; This is in the form of a MAC address.
|
||||
; It should be universally unique.
|
||||
; It must be unique between servers communicating
|
||||
; with a protocol that uses this value.
|
||||
; This is currently is used by DUNDi and
|
||||
; Exchanging Device and Mailbox State
|
||||
; using protocols: XMPP, Corosync and PJSIP.
|
||||
;rtp_use_dynamic = yes ; When set to "yes" RTP dynamic payload types
|
||||
; are assigned dynamically per RTP instance vs.
|
||||
; allowing Asterisk to globally initialize them
|
||||
; to pre-designated numbers (defaults to "yes").
|
||||
;rtp_pt_dynamic = 35 ; Normally the Dynamic RTP Payload Type numbers
|
||||
; are 96-127, which allow just 32 formats. The
|
||||
; starting point 35 enables the range 35-63 and
|
||||
; allows 29 additional formats. When you use
|
||||
; more than 32 formats in the dynamic range and
|
||||
; calls are not accepted by a remote
|
||||
; implementation, please report this and go
|
||||
; back to value 96.
|
||||
;hide_messaging_ami_events = no; This option, if enabled, will
|
||||
; suppress all of the Message/ast_msg_queue channel's
|
||||
; housekeeping AMI and ARI channel events. This can
|
||||
; reduce the load on the manager and ARI applications
|
||||
; when the Digium Phone Module for Asterisk is in use.
|
||||
|
||||
; Changing the following lines may compromise your security.
|
||||
;[files]
|
||||
;astctlpermissions = 0660
|
||||
;astctlowner = root
|
||||
;astctlgroup = apache
|
||||
;astctl = asterisk.ctl
|
||||
111
asterisk/calendar.conf
Executable file
111
asterisk/calendar.conf
Executable file
@ -0,0 +1,111 @@
|
||||
;[calendar1]
|
||||
;type = ical ; type of calendar--currently supported: ical, caldav, exchange, or ews
|
||||
;url = https://example.com/home/jdoe/Calendar/ ; URL to shared calendar (Zimbra example)
|
||||
;user = jdoe ; web username
|
||||
;secret = supersecret ; web password
|
||||
;refresh = 15 ; refresh calendar every n minutes
|
||||
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
|
||||
; ; should always be >= refresh
|
||||
;fetch_again_at_reload = no ; to reload the calendar content when the module is reloaded
|
||||
;
|
||||
;
|
||||
; You can set up res_calendar to execute a call upon an upcoming busy status
|
||||
; The following fields are available from the ${CALENDAR_EVENT(<field>)} dialplan function:
|
||||
;
|
||||
; summary : The VEVENT Summary property or Exchange subject
|
||||
; description : The text description of the vent
|
||||
; organizer : The organizer of the event
|
||||
; location : The location field of the event
|
||||
; calendar : The name of the calendar tied to the event
|
||||
; uid : The unique ID for this event
|
||||
; start : Start time of the event
|
||||
; end : The end time of the event
|
||||
; busystate : 0=FREE, 1=TENTATIVE, 2=BUSY
|
||||
;
|
||||
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
|
||||
;
|
||||
;channel = SIP/60001 ; Channel to dial
|
||||
;context = default ; Context to connect to on answer
|
||||
;extension = 123 ; Extension to connect to on answer
|
||||
;
|
||||
; or
|
||||
;
|
||||
;app = Playback ; Application to execute on answer (instead of context/extension)
|
||||
;appdata = tt-weasels ; Data part of application to execute on answer
|
||||
;
|
||||
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
|
||||
;
|
||||
; Channel variables can be set on the notification channel. The format is
|
||||
; setvar=name=value. Variable subsitution is done on the value to allow the use of dialplan
|
||||
; functions like CALENDAR_EVENT. The variables are set in order, so one can use the value
|
||||
; of earlier variables in the definition of later ones.
|
||||
;
|
||||
;setvar = CALLERID(name)=${CALENDAR_EVENT(summary)}
|
||||
|
||||
;[calendar2]
|
||||
; Note: Support for Exchange Server 2003
|
||||
;
|
||||
;type = exchange ; type of calendar--currently supported: ical, caldav, exchange, or ews
|
||||
;url = https://example.com/exchange/jdoe ; URL to MS Exchange OWA for user (usually includes exchange/user)
|
||||
;user = jdoe ; Exchange username
|
||||
;secret = mysecret ; Exchange password
|
||||
;refresh = 15 ; refresh calendar every n minutes
|
||||
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
|
||||
; ; should always be >= refresh
|
||||
;
|
||||
; You can set up res_calendar to execute a call upon an upcoming busy status
|
||||
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
|
||||
;
|
||||
;channel = SIP/1234 ; Channel to dial
|
||||
;context = default ; Context to connect to on answer
|
||||
;extension = 1234 ; Extension to connect to on answer
|
||||
;
|
||||
; or
|
||||
;
|
||||
;[calendar3]
|
||||
; Note: Support for Exchange Server 2007+
|
||||
;
|
||||
;type = ews ; type of calendar--currently supported: ical, caldav, exchange, or ews
|
||||
;url = https://example.com/ews/Exchange.asmx ; URL to MS Exchange EWS
|
||||
;user = jdoe ; Exchange username
|
||||
;secret = mysecret ; Exchange password
|
||||
;refresh = 15 ; refresh calendar every n minutes
|
||||
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
|
||||
; ; should always be >= refresh
|
||||
;
|
||||
; You can set up res_calendar to execute a call upon an upcoming busy status
|
||||
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
|
||||
;
|
||||
;channel = SIP/1234 ; Channel to dial
|
||||
;context = default ; Context to connect to on answer
|
||||
;extension = 1234 ; Extension to connect to on answer
|
||||
;
|
||||
; or
|
||||
;
|
||||
;app = Playback ; Application to execute on answer (instead of context/extension)
|
||||
;appdata = tt-weasels ; Data part of application to execute on answer
|
||||
;
|
||||
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
|
||||
|
||||
;[calendar4]
|
||||
;type = caldav ; type of calendar--currently supported: ical, caldav, exchange, or ews
|
||||
;url = https://www.google.com/calendar/dav/username@gmail.com/events/ ; Main GMail calendar (the trailing slash is significant!)
|
||||
;user = jdoe@gmail.com ; username
|
||||
;secret = mysecret ; password
|
||||
;refresh = 15 ; refresh calendar every n minutes
|
||||
;timeframe = 60 ; number of minutes of calendar data to pull for each refresh period
|
||||
; ; should always be >= refresh
|
||||
;
|
||||
; You can set up res_calendar to execute a call upon an upcoming busy status
|
||||
;autoreminder = 10 ; Override event-defined reminder before each busy status (in mins)
|
||||
;
|
||||
;channel = SIP/1234 ; Channel to dial
|
||||
;context = default ; Context to connect to on answer
|
||||
;extension = 1234 ; Extension to connect to on answer
|
||||
;
|
||||
; or
|
||||
;
|
||||
;app = Playback ; Application to execute on answer (instead of context/extension)
|
||||
;appdata = tt-weasels ; Data part of application to execute on answer
|
||||
;
|
||||
;waittime = 30 ; How long to wait for an answer, defaults to 30 seconds
|
||||
205
asterisk/ccss.conf
Executable file
205
asterisk/ccss.conf
Executable file
@ -0,0 +1,205 @@
|
||||
;
|
||||
; --- Call Completion Supplementary Services ---
|
||||
;
|
||||
; For more information about CCSS, see the CCSS user documentation
|
||||
; https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+(CCSS)
|
||||
;
|
||||
|
||||
[general]
|
||||
; The cc_max_requests option is a global limit on the number of
|
||||
; CC requests that may be in the Asterisk system at any time.
|
||||
;
|
||||
;cc_max_requests = 20
|
||||
;
|
||||
; The cc_STATE_devstate variables listed below can be used to change the
|
||||
; default mapping of the internal state machine tracking the state of
|
||||
; call completion to an Asterisk Device State value. The acceptable values
|
||||
; that can be provided are as follows, with a description of what the
|
||||
; equivalent device BLF that this maps to:
|
||||
;
|
||||
; UNKNOWN ; Device is valid but channel didn't know state
|
||||
; NOT_INUSE ; Device is not used
|
||||
; INUSE ; Device is in use
|
||||
; BUSY ; Device is busy
|
||||
; INVALID ; Device is invalid
|
||||
; UNAVAILABLE ; Device is unavailable
|
||||
; RINGING ; Device is ringing
|
||||
; RINGINUSE ; Device is ringing *and* in use
|
||||
; ONHOLD ; Device is on hold
|
||||
;
|
||||
; These states are used to generate DEVICE_STATE information that can be
|
||||
; included with Asterisk hints for phones to subscribe to the state information
|
||||
; or dialplan to check the state using the EXTENSION_STATE() function or
|
||||
; the DEVICE_STATE() function.
|
||||
;
|
||||
; The states are in the format of: "ccss:TECH/ID" so an example of device
|
||||
; SIP/3000 making a CallCompletionRequest() could be checked by looking at
|
||||
; DEVICE_STATE(ccss:SIP/3000) or an Asterisk Hint could be generated such as
|
||||
;
|
||||
; [hint-context]
|
||||
; exten => *843000,hint,ccss:SIP/3000
|
||||
;
|
||||
; and then accessed with EXTENSION_STATE(*843000@hint-context)
|
||||
; or subscribed to with a BLF button on a phone.
|
||||
;
|
||||
; The available state mapping and default values are:
|
||||
;
|
||||
; cc_available_devstate = NOT_INUSE
|
||||
; cc_offered_devstate = NOT_INUSE
|
||||
; cc_caller_requested_devstate = NOT_INUSE
|
||||
; cc_active_devstate = INUSE
|
||||
; cc_callee_ready_devstate = INUSE
|
||||
; cc_caller_busy_devstate = ONHOLD
|
||||
; cc_recalling_devstate = RINGING
|
||||
; cc_complete_devstate = NOT_INUSE
|
||||
; cc_failed_devstate = NOT_INUSE
|
||||
|
||||
;
|
||||
;============================================
|
||||
; PLEASE READ THIS!!!
|
||||
; The options described below should NOT be
|
||||
; set in this file. Rather, they should be
|
||||
; set per-device in a channel driver
|
||||
; configuration file.
|
||||
; PLEASE READ THIS!!!
|
||||
;===========================================
|
||||
;
|
||||
; --------------------------------------------------------------------
|
||||
; Timers
|
||||
; --------------------------------------------------------------------
|
||||
;There are three configurable timers for all types of CC: the
|
||||
;cc_offer_timer, the ccbs_available_timer, and the ccnr_available_timer.
|
||||
;In addition, when using a generic agent, there is a fourth timer,
|
||||
;the cc_recall_timer. All timers are configured in seconds, and the
|
||||
;values shown below are the defaults.
|
||||
;
|
||||
;When a caller is offered CCBS or CCNR, the cc_offer_timer will
|
||||
;be started. If the caller does not request CC before the
|
||||
;cc_offer_timer expires, then the caller will be unable to request
|
||||
;CC for this call.
|
||||
;
|
||||
;cc_offer_timer = 20
|
||||
;
|
||||
;Once a caller has requested CC, then either the ccbs_available_timer
|
||||
;or the ccnr_available_timer will run, depending on the service
|
||||
;requested. The reason why there are two separate timers for CCBS
|
||||
;and CCNR is that it is reasonable to want to have a shorter timeout
|
||||
;configured for CCBS than for CCNR. If the available timer expires
|
||||
;before the called party becomes available, then the CC attempt
|
||||
;will have failed and monitoring of the called party will stop.
|
||||
;
|
||||
;ccbs_available_timer = 4800
|
||||
;ccnr_available_timer = 7200
|
||||
;
|
||||
; When using a generic agent, the original caller is called back
|
||||
; when one of the original called parties becomes available. The
|
||||
; cc_recall_timer tells Asterisk how long it should let the original
|
||||
; caller's phone ring before giving up. Please note that this parameter
|
||||
; only affects operation when using a generic agent.
|
||||
;
|
||||
;cc_recall_timer = 20
|
||||
; --------------------------------------------------------------------
|
||||
; Policies
|
||||
; --------------------------------------------------------------------
|
||||
; Policy settings tell Asterisk how to behave and what sort of
|
||||
; resources to allocate in order to facilitate CC. There are two
|
||||
; settings to control the actions Asterisk will take.
|
||||
;
|
||||
; The cc_agent_policy describes the behavior that Asterisk will
|
||||
; take when communicating with the caller during CC. There are
|
||||
; three possible options.
|
||||
;
|
||||
;never: Never offer CC to the caller. Setting the cc_agent_policy
|
||||
; to this value is the way to disable CC for a call.
|
||||
;
|
||||
;generic: A generic CC agent is one which uses no protocol-specific
|
||||
; mechanisms to offer CC to the caller. Instead, the caller
|
||||
; requests CC using a dialplan function. Due to internal
|
||||
; restrictions, you should only use a generic CC agent on
|
||||
; phones (i.e. not "trunks"). If you are using phones which
|
||||
; do not support a protocol-specific method of using CC, then
|
||||
; generic CC agents are what you should use.
|
||||
;
|
||||
;native: A native CC agent is one which uses protocol-specific
|
||||
; signaling to offer CC to the caller and accept CC requests
|
||||
; from the caller. The supported protocols for native CC
|
||||
; agents are SIP, ISDN ETSI PTP, ISDN ETSI PTMP, and Q.SIG
|
||||
;cc_agent_policy=never
|
||||
;
|
||||
; The cc_monitor_policy describes the behavior that Asterisk will
|
||||
; take when communicating with the called party during CC. There
|
||||
; are four possible options.
|
||||
;
|
||||
;never: Analogous to the cc_agent_policy setting. We will never
|
||||
; attempt to request CC services on this interface.
|
||||
;
|
||||
;generic: Analogous to the cc_agent_policy setting. We will monitor
|
||||
; the called party's progress using protocol-agnostic
|
||||
; capabilities. Like with generic CC agents, generic CC
|
||||
; monitors should only be used for phones.
|
||||
;
|
||||
;native: Analogous to the cc_agent_policy setting. We will use
|
||||
; protocol-specific methods to request CC from this interface
|
||||
; and to monitor the interface for availability.
|
||||
;
|
||||
;always: If an interface is set to "always," then we will accept
|
||||
; protocol-specific CC offers from the caller and use
|
||||
; a native CC monitor for the remainder of the CC transaction.
|
||||
; However, if the interface does not offer protocol-specific
|
||||
; CC, then we will fall back to using a generic CC monitor
|
||||
; instead. This is a good setting to use for phones for which
|
||||
; you do not know if they support protocol-specific CC
|
||||
; methodologies.
|
||||
;cc_monitor_policy=never
|
||||
;
|
||||
;
|
||||
; --------------------------------------------------------------------
|
||||
; Limits
|
||||
; --------------------------------------------------------------------
|
||||
;
|
||||
; The use of CC requires Asterisk to potentially use more memory than
|
||||
; some administrators would like. As such, it is a good idea to limit
|
||||
; the number of CC requests that can be in the system at a given time.
|
||||
; The values shown below are the defaults.
|
||||
;
|
||||
; The cc_max_agents setting limits the number of outstanding CC
|
||||
; requests a caller may have at any given time. Please note that due
|
||||
; to implementation restrictions, this setting is ignored when using
|
||||
; generic CC agents. Generic CC agents may only have one outstanding
|
||||
; CC request.
|
||||
;
|
||||
;cc_max_agents = 5
|
||||
;
|
||||
; The cc_max_monitors setting limits the number of outstanding CC
|
||||
; requests can be made to a specific interface at a given time.
|
||||
;
|
||||
;cc_max_monitors = 5
|
||||
;
|
||||
; --------------------------------------------------------------------
|
||||
; Other
|
||||
; --------------------------------------------------------------------
|
||||
;
|
||||
; When using a generic CC agent, the caller who requested CC will be
|
||||
; called back when a called party becomes available. When the caller
|
||||
; answers his phone, the administrator may opt to have a macro run.
|
||||
; What this macro does is up to the administrator. By default there
|
||||
; is no callback macro configured.
|
||||
;
|
||||
;cc_callback_macro=
|
||||
;
|
||||
; Alternatively, the administrator may run a subroutine. By default
|
||||
; there is no callback subroutine configured. The subroutine should
|
||||
; be specified in the format: [[context,]exten,]priority
|
||||
;
|
||||
;cc_callback_sub=
|
||||
;
|
||||
; When using an ISDN phone and a generic CC agent, Asterisk is unable
|
||||
; to determine the dialstring that should be used when calling back
|
||||
; the original caller. Furthermore, if you desire to use any dialstring-
|
||||
; specific options, such as distinctive ring, you must set this
|
||||
; configuration option. For non-ISDN phones, it is not necessary to
|
||||
; set this, since Asterisk can determine the dialstring to use since
|
||||
; it is identical to the name of the calling device. By default, there
|
||||
; is no cc_agent_dialstring set.
|
||||
;
|
||||
;cc_agent_dialstring=
|
||||
167
asterisk/cdr.conf
Executable file
167
asterisk/cdr.conf
Executable file
@ -0,0 +1,167 @@
|
||||
;
|
||||
; Asterisk Call Detail Record engine configuration
|
||||
;
|
||||
; CDR is Call Detail Record, which provides logging services via a variety of
|
||||
; pluggable backend modules. Detailed call information can be recorded to
|
||||
; databases, files, etc. Useful for billing, fraud prevention, compliance with
|
||||
; Sarbanes-Oxley aka The Enron Act, QOS evaluations, and more.
|
||||
;
|
||||
|
||||
[general]
|
||||
|
||||
; Define whether or not to use CDR logging. Setting this to "no" will override
|
||||
; any loading of backend CDR modules. Default is "yes".
|
||||
;enable=yes
|
||||
|
||||
; Define whether or not to log unanswered calls that don't involve an outgoing
|
||||
; party. Setting this to "yes" will make calls to extensions that don't answer
|
||||
; and don't set a B side channel (such as by using the Dial application)
|
||||
; receive CDR log entries. If this option is set to "no", then those log
|
||||
; entries will not be created. Unanswered Calls which get offered to an
|
||||
; outgoing line will always receive log entries regardless of this option, and
|
||||
; that is the intended behaviour.
|
||||
;unanswered = no
|
||||
|
||||
; Define whether or not to log congested calls. Setting this to "yes" will
|
||||
; report each call that fails to complete due to congestion conditions. Default
|
||||
; is "no".
|
||||
;congestion = no
|
||||
|
||||
; Normally, CDR's are not closed out until after all extensions are finished
|
||||
; executing. By enabling this option, the CDR will be ended before executing
|
||||
; the "h" extension and hangup handlers so that CDR values such as "end" and
|
||||
; "billsec" may be retrieved inside of of this extension.
|
||||
; The default value is "no".
|
||||
;endbeforehexten=no
|
||||
|
||||
; Normally, the 'billsec' field logged to the backends (text files or databases)
|
||||
; is simply the end time (hangup time) minus the answer time in seconds. Internally,
|
||||
; asterisk stores the time in terms of microseconds and seconds. By setting
|
||||
; initiatedseconds to 'yes', you can force asterisk to report any seconds
|
||||
; that were initiated (a sort of round up method). Technically, this is
|
||||
; when the microsecond part of the end time is greater than the microsecond
|
||||
; part of the answer time, then the billsec time is incremented one second.
|
||||
; The default value is "no".
|
||||
;initiatedseconds=no
|
||||
|
||||
; Define the CDR batch mode, where instead of posting the CDR at the end of
|
||||
; every call, the data will be stored in a buffer to help alleviate load on the
|
||||
; asterisk server. Default is "no".
|
||||
;
|
||||
; WARNING WARNING WARNING
|
||||
; Use of batch mode may result in data loss after unsafe asterisk termination
|
||||
; ie. software crash, power failure, kill -9, etc.
|
||||
; WARNING WARNING WARNING
|
||||
;
|
||||
;batch=no
|
||||
|
||||
; Define the maximum number of CDRs to accumulate in the buffer before posting
|
||||
; them to the backend engines. 'batch' must be set to 'yes'. Default is 100.
|
||||
;size=100
|
||||
|
||||
; Define the maximum time to accumulate CDRs in the buffer before posting them
|
||||
; to the backend engines. If this time limit is reached, then it will post the
|
||||
; records, regardless of the value defined for 'size'. 'batch' must be set to
|
||||
; 'yes'. Note that time is in seconds. Default is 300 (5 minutes).
|
||||
;time=300
|
||||
|
||||
; The CDR engine uses the internal asterisk scheduler to determine when to post
|
||||
; records. Posting can either occur inside the scheduler thread, or a new
|
||||
; thread can be spawned for the submission of every batch. For small batches,
|
||||
; it might be acceptable to just use the scheduler thread, so set this to "yes".
|
||||
; For large batches, say anything over size=10, a new thread is recommended, so
|
||||
; set this to "no". Default is "no".
|
||||
;scheduleronly=no
|
||||
|
||||
; When shutting down asterisk, you can block until the CDRs are submitted. If
|
||||
; you don't, then data will likely be lost. You can always check the size of
|
||||
; the CDR batch buffer with the CLI "cdr status" command. To enable blocking on
|
||||
; submission of CDR data during asterisk shutdown, set this to "yes". Default
|
||||
; is "yes".
|
||||
;safeshutdown=yes
|
||||
|
||||
;
|
||||
;
|
||||
; CHOOSING A CDR "BACKEND" (what kind of output to generate)
|
||||
;
|
||||
; To choose a backend, you have to make sure either the right category is
|
||||
; defined in this file, or that the appropriate config file exists, and has the
|
||||
; proper definitions in it. If there are any problems, usually, the entry will
|
||||
; silently ignored, and you get no output.
|
||||
;
|
||||
; Also, please note that you can generate CDR records in as many formats as you
|
||||
; wish. If you configure 5 different CDR formats, then each event will be logged
|
||||
; in 5 different places! In the example config files, all formats are commented
|
||||
; out except for the cdr-csv format.
|
||||
;
|
||||
; Here are all the possible back ends:
|
||||
;
|
||||
; csv, custom, manager, odbc, pgsql, radius, sqlite, tds
|
||||
; (also, mysql is available via the asterisk-addons, due to licensing
|
||||
; requirements)
|
||||
; (please note, also, that other backends can be created, by creating
|
||||
; a new backend module in the source cdr/ directory!)
|
||||
;
|
||||
; Some of the modules required to provide these backends will not build or install
|
||||
; unless some dependency requirements are met. Examples of this are pgsql, odbc,
|
||||
; etc. If you are not getting output as you would expect, the first thing to do
|
||||
; is to run the command "make menuselect", and check what modules are available,
|
||||
; by looking in the "2. Call Detail Recording" option in the main menu. If your
|
||||
; backend is marked with XXX, you know that the "configure" command could not find
|
||||
; the required libraries for that option.
|
||||
;
|
||||
; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
|
||||
; file, define the [csv] category in this file. No database necessary. The example
|
||||
; config files are set up to provide this kind of output by default.
|
||||
;
|
||||
; To get custom csv CDR records, make sure the cdr_custom.conf file
|
||||
; is present, and contains the proper [mappings] section. The advantage to
|
||||
; using this backend, is that you can define which fields to output, and in
|
||||
; what order. By default, the example configs are set up to mimic the cdr-csv
|
||||
; output. If you don't make any changes to the mappings, you are basically generating
|
||||
; the same thing as cdr-csv, but expending more CPU cycles to do so!
|
||||
;
|
||||
; To get manager events generated, make sure the cdr_manager.conf file exists,
|
||||
; and the [general] section is defined, with the single variable 'enabled = yes'.
|
||||
;
|
||||
; For odbc, make sure all the proper libs are installed, that "make menuselect"
|
||||
; shows that the modules are available, and the cdr_odbc.conf file exists, and
|
||||
; has a [global] section with the proper variables defined.
|
||||
;
|
||||
; For pgsql, make sure all the proper libs are installed, that "make menuselect"
|
||||
; shows that the modules are available, and the cdr_pgsql.conf file exists, and
|
||||
; has a [global] section with the proper variables defined.
|
||||
;
|
||||
; For logging to radius databases, make sure all the proper libs are installed, that
|
||||
; "make menuselect" shows that the modules are available, and the [radius]
|
||||
; category is defined in this file, and in that section, make sure the 'radiuscfg'
|
||||
; variable is properly pointing to an existing radiusclient.conf file.
|
||||
;
|
||||
; For logging to sqlite databases, make sure the 'cdr.db' file exists in the log directory,
|
||||
; which is usually /var/log/asterisk. Of course, the proper libraries should be available
|
||||
; during the 'configure' operation.
|
||||
;
|
||||
; For tds logging, make sure the proper libraries are available during the 'configure'
|
||||
; phase, and that cdr_tds.conf exists and is properly set up with a [global] category.
|
||||
;
|
||||
; Also, remember, that if you wish to log CDR info to a database, you will have to define
|
||||
; a specific table in that databse to make things work! See the doc directory for more details
|
||||
; on how to create this table in each database.
|
||||
;
|
||||
|
||||
[csv]
|
||||
usegmtime=yes ; log date/time in GMT. Default is "no"
|
||||
loguniqueid=yes ; log uniqueid. Default is "no"
|
||||
loguserfield=yes ; log user field. Default is "no"
|
||||
accountlogs=yes ; create separate log file for each account code. Default is "yes"
|
||||
;newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns (peeraccount, linkedid, sequence).
|
||||
; Default is "no".
|
||||
|
||||
;[radius]
|
||||
;usegmtime=yes ; log date/time in GMT
|
||||
;loguniqueid=yes ; log uniqueid
|
||||
;loguserfield=yes ; log user field
|
||||
; Set this to the location of the radiusclient-ng configuration file
|
||||
; The default is /etc/radiusclient-ng/radiusclient.conf
|
||||
;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
|
||||
radiuscfg => /etc/radcli/radiusclient.conf
|
||||
62
asterisk/cdr_adaptive_odbc.conf
Executable file
62
asterisk/cdr_adaptive_odbc.conf
Executable file
@ -0,0 +1,62 @@
|
||||
; The point of this module is to allow you log whatever you like in terms of
|
||||
; the CDR variables. Do you want to log uniqueid? Then simply ensure that
|
||||
; your table has that column. If you don't want the column, ensure that it
|
||||
; does not exist in the table structure. If you'd like to call uniqueid
|
||||
; something else in your table, simply provide an alias in the configuration
|
||||
; file that maps the standard CDR field name (uniqueid) to whatever column
|
||||
; name you like. Perhaps you'd like some extra CDR values logged that aren't
|
||||
; in the standard repertoire of CDR variables (some that come to mind are
|
||||
; certain values used for LCR: route, per_minute_cost, and per_minute_price).
|
||||
; Simply set those CDR variables in your dialplan, i.e. Set(CDR(route)=27),
|
||||
; ensure that a corresponding column exists in your table, and cdr_adaptive_odbc
|
||||
; will do the rest.
|
||||
;
|
||||
; This configuration defines the connections and tables for which CDRs may
|
||||
; be populated. Each context specifies a different CDR table to be used.
|
||||
;
|
||||
; The columns in the tables should match up word-for-word (case-insensitive)
|
||||
; to the CDR variables set in the dialplan. The natural advantage to this
|
||||
; system is that beyond setting up the configuration file to tell you what
|
||||
; tables to look at, there isn't anything more to do beyond creating the
|
||||
; columns for the fields that you want, and populating the corresponding
|
||||
; CDR variables in the dialplan. For the builtin variables only, you may
|
||||
; create aliases for the real column name.
|
||||
;
|
||||
; Please note that after adding columns to the database, it is necessary to
|
||||
; reload this module to get the new column names and types read.
|
||||
;
|
||||
; Warning: if you specify two contexts with exactly the same connection and
|
||||
; table names, you will get duplicate records in that table. So be careful.
|
||||
;
|
||||
|
||||
;[first]
|
||||
;connection=mysql1
|
||||
;table=cdr
|
||||
|
||||
;[second]
|
||||
;connection=mysql1
|
||||
;table=extracdr
|
||||
|
||||
;[third]
|
||||
;connection=sqlserver
|
||||
;table=AsteriskCDR
|
||||
;schema=public ; for databases which support schemas
|
||||
;usegmtime=yes ; defaults to no
|
||||
;alias src => source
|
||||
;alias channel => source_channel
|
||||
;alias dst => dest
|
||||
;alias dstchannel => dest_channel
|
||||
;
|
||||
; Any filter specified MUST match exactly or the CDR will be discarded
|
||||
;filter accountcode => somename
|
||||
;filter src => 123
|
||||
; Negative filters are also now available
|
||||
;filter src != 456
|
||||
;
|
||||
; Additionally, we now support setting static values per column. The reason
|
||||
; for this is to allow different sections to specify different values for
|
||||
; a certain named column, presumably separated by filters.
|
||||
;static "Some Special Value" => identifier_code
|
||||
;
|
||||
; Add quoted indentifiers for table and columns names.
|
||||
;quoted_identifiers=" ; Default to null
|
||||
20
asterisk/cdr_beanstalkd.conf
Executable file
20
asterisk/cdr_beanstalkd.conf
Executable file
@ -0,0 +1,20 @@
|
||||
;
|
||||
; Asterisk Call Management CDR via Beanstalkd job queue
|
||||
;
|
||||
; Beanstalkd is a simple job queue server, that is highly versatile and simple to use.
|
||||
; Beanstalkd includes the capability of using multiple queues at the same time, with priorities.
|
||||
;
|
||||
; This module requires that your server has the beanstalk-client library installed. The library
|
||||
; can be downloaded from - https://github.com/deepfryed/beanstalk-client
|
||||
;
|
||||
|
||||
[general]
|
||||
;enabled = yes
|
||||
|
||||
;host = 127.0.0.1 ; Specify the remote IP address of the Beanstalkd server
|
||||
;port = 11300 ; Specify the remote PORT of the the Beanstalkd server
|
||||
;tube = asterisk-cdr ; Specify the default CDR job queue to use
|
||||
;priority = 99 ; Specify the default job priority for the queue. This parameter is useful when building
|
||||
; platform with multiple Asterisk servers, that are used for different functions. For example,
|
||||
; none billable CDR records can be inserted with a priority of 99, while billable ones be
|
||||
; inserted with a priority of 1
|
||||
14
asterisk/cdr_custom.conf
Executable file
14
asterisk/cdr_custom.conf
Executable file
@ -0,0 +1,14 @@
|
||||
;
|
||||
; Mappings for custom config file
|
||||
;
|
||||
; To get your CSV output in a format tailored to your liking, uncomment the
|
||||
; following lines and look for the output in the cdr-custom directory (usually
|
||||
; in /var/log/asterisk). Depending on which mapping you uncomment, you may see
|
||||
; Master.csv, Simple.csv, or both.
|
||||
;
|
||||
;[mappings]
|
||||
;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
|
||||
;
|
||||
; High Resolution Time for billsec and duration fields
|
||||
;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
|
||||
;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}
|
||||
15
asterisk/cdr_manager.conf
Executable file
15
asterisk/cdr_manager.conf
Executable file
@ -0,0 +1,15 @@
|
||||
;
|
||||
; Asterisk Call Management CDR
|
||||
;
|
||||
[general]
|
||||
enabled = no
|
||||
|
||||
; The "mappings" category can be used to define additional "key: value" pairs
|
||||
; that will be included in the manager event. (after AccountCode, Source, etc).
|
||||
;
|
||||
; Each line like "varname => label" will include a "label: ${CDR(varname)}"
|
||||
; in the generated event where ${CDR(varname)} its replaced with its value
|
||||
;
|
||||
;[mappings]
|
||||
;rate => Rate
|
||||
;carrier => Carrier
|
||||
62
asterisk/cdr_mysql.conf
Executable file
62
asterisk/cdr_mysql.conf
Executable file
@ -0,0 +1,62 @@
|
||||
;
|
||||
; Note - if the database server is hosted on the same machine as the
|
||||
; asterisk server, you can achieve a local Unix socket connection by
|
||||
; setting hostname=localhost
|
||||
;
|
||||
; port and sock are both optional parameters. If hostname is specified
|
||||
; and is not "localhost" (you can use address 127.0.0.1 instead), then
|
||||
; cdr_mysql will attempt to connect to the port specified or use the
|
||||
; default port. If hostname is not specified or if hostname is
|
||||
; "localhost", then cdr_mysql will attempt to connect to the socket file
|
||||
; specified by sock or otherwise use the default socket file.
|
||||
;
|
||||
;[global]
|
||||
;hostname=database.host.name
|
||||
;dbname=asteriskcdrdb
|
||||
;table=cdr
|
||||
;password=password
|
||||
;user=asteriskcdruser
|
||||
;port=3306
|
||||
;sock=/tmp/mysql.sock
|
||||
; By default CDRs are logged in the system's time zone
|
||||
;cdrzone=UTC ; log CDRs with UTC
|
||||
;usegmtime=yes ;log date/time in GMT. Default is "no"
|
||||
;cdrzone=America/New_York ; or use a specific time zone
|
||||
;
|
||||
; If your system's locale differs from mysql database character set,
|
||||
; cdr_mysql can damage non-latin characters in CDR variables. Use this
|
||||
; option to protect your data.
|
||||
;charset=koi8r
|
||||
;
|
||||
; Older versions of cdr_mysql set the calldate field to whenever the
|
||||
; record was posted, rather than the start date of the call. This flag
|
||||
; reverts to the old (incorrect) behavior. Note that you'll also need
|
||||
; to comment out the "start=calldate" alias, below, to use this.
|
||||
;compat=no
|
||||
;
|
||||
; ssl connections (optional)
|
||||
;ssl_ca=<path to CA cert>
|
||||
;ssl_cert=<path to cert>
|
||||
;ssl_key=<path to keyfile>
|
||||
;
|
||||
; You may also configure the field names used in the CDR table.
|
||||
;
|
||||
[columns]
|
||||
;static "<value>" => <column>
|
||||
;alias <cdrvar> => <column>
|
||||
alias start => calldate
|
||||
;alias clid => <a_field_not_named_clid>
|
||||
;alias src => <a_field_not_named_src>
|
||||
;alias dst => <a_field_not_named_dst>
|
||||
;alias dcontext => <a_field_not_named_dcontext>
|
||||
;alias channel => <a_field_not_named_channel>
|
||||
;alias dstchannel => <a_field_not_named_dstchannel>
|
||||
;alias lastapp => <a_field_not_named_lastapp>
|
||||
;alias lastdata => <a_field_not_named_lastdata>
|
||||
;alias duration => <a_field_not_named_duration>
|
||||
;alias billsec => <a_field_not_named_billsec>
|
||||
;alias disposition => <a_field_not_named_disposition>
|
||||
;alias amaflags => <a_field_not_named_amaflags>
|
||||
;alias accountcode => <a_field_not_named_accountcode>
|
||||
;alias userfield => <a_field_not_named_userfield>
|
||||
;alias uniqueid => <a_field_not_named_uniqueid>
|
||||
12
asterisk/cdr_odbc.conf
Executable file
12
asterisk/cdr_odbc.conf
Executable file
@ -0,0 +1,12 @@
|
||||
;
|
||||
; cdr_odbc.conf
|
||||
;
|
||||
|
||||
;[global]
|
||||
;dsn=MySQL-test
|
||||
;loguniqueid=yes
|
||||
;dispositionstring=yes
|
||||
;table=cdr ;"cdr" is default table name
|
||||
;usegmtime=no ; set to "yes" to log in GMT
|
||||
;hrtime=yes ;Enables microsecond accuracy with the billsec and duration fields
|
||||
;newcdrcolumns=yes ; Enable logging of post-1.8 CDR columns (peeraccount, linkedid, sequence)
|
||||
16
asterisk/cdr_pgsql.conf
Executable file
16
asterisk/cdr_pgsql.conf
Executable file
@ -0,0 +1,16 @@
|
||||
; Sample Asterisk config file for CDR logging to PostgresSQL
|
||||
;
|
||||
; Note that you can use TLS connections to your database server.
|
||||
; This is configured for your PostgreSQL client installation
|
||||
; on this system (check for pgsessions.conf)
|
||||
|
||||
[global]
|
||||
;hostname=localhost
|
||||
;port=5432
|
||||
;dbname=asterisk
|
||||
;password=password
|
||||
;user=postgres
|
||||
;appname=asterisk ; Postgres application_name support (optional). Whitespace not allowed.
|
||||
;table=cdr ;SQL table where CDRs will be inserted
|
||||
;encoding=LATIN9 ; Encoding of logged characters in Asterisk
|
||||
;timezone=UTC ; Uncomment if you want datetime fields in UTC/GMT
|
||||
11
asterisk/cdr_sqlite3_custom.conf
Executable file
11
asterisk/cdr_sqlite3_custom.conf
Executable file
@ -0,0 +1,11 @@
|
||||
;
|
||||
; Mappings for custom config file
|
||||
;
|
||||
[master] ; currently, only file "master.db" is supported, with only one table at a time.
|
||||
;table => cdr
|
||||
;columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield, test
|
||||
;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}','${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'
|
||||
;busy_timeout => 1000
|
||||
|
||||
;Enable High Resolution Times for billsec and duration fields
|
||||
;values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration,f)}','${CDR(billsec,f)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}','${CDR(test)}'
|
||||
83
asterisk/cdr_syslog.conf
Executable file
83
asterisk/cdr_syslog.conf
Executable file
@ -0,0 +1,83 @@
|
||||
;
|
||||
; Asterisk Call Detail Records (CDR) - Syslog Backend
|
||||
;
|
||||
|
||||
; The cdr_syslog module writes CDRs using the facilities provided by syslog.
|
||||
;
|
||||
; Not only must you configure cdr_syslog from this file (cdr_syslog.conf) but
|
||||
; you will also need to make changes to your /etc/syslog.conf before CDRs will
|
||||
; be written to syslog.
|
||||
;
|
||||
; As an example, you can add the following to /etc/syslog.conf:
|
||||
;
|
||||
; local4.info /var/log/asterisk-cdr.log
|
||||
;
|
||||
; And then instruct syslogd to re-read the configuration file by sending it a
|
||||
; HUP signal. On Linux this can be done like this:
|
||||
;
|
||||
; kill -HUP `cat /var/run/syslogd.pid`
|
||||
;
|
||||
; Finally, you will need to uncomment the [cdr-simple] section below, and restart
|
||||
; Asterisk. When calls are placed, you should start seeing records appear in
|
||||
; /var/log/asterisk-cdr.log.
|
||||
|
||||
[general]
|
||||
; Facility
|
||||
;
|
||||
; The 'facility' keyword specifies the syslog facility to use when writing out
|
||||
; CDRs.
|
||||
;
|
||||
; Accepted values: One of the following:
|
||||
; user, local0, local1, local2, local3, local4, local5, local6
|
||||
; and local7.
|
||||
;
|
||||
; Note: Depending on your platform, the following may also be
|
||||
; available:
|
||||
; auth, authpriv, cron, daemon, ftp, kern, lpr, mail,
|
||||
; news, syslog, and uucp.
|
||||
;
|
||||
; Default value: local4
|
||||
|
||||
;facility=local0
|
||||
|
||||
; Priority
|
||||
;
|
||||
; Use the 'priority' keyword to select which of the syslog priority levels to
|
||||
; use when logging CDRs.
|
||||
;
|
||||
; Accepted values: One of the following:
|
||||
; alert, crit, debug, emerg, err, info, notice, warning
|
||||
; Default value: info
|
||||
|
||||
;priority=warn
|
||||
|
||||
; Note: The settings for 'facility' and 'priority' in the [general] section
|
||||
; define the default values for all of the logging locations created
|
||||
; below in separate sections.
|
||||
|
||||
;[cdr-master]
|
||||
;facility = local5
|
||||
;priority = debug
|
||||
|
||||
; Template
|
||||
;
|
||||
; The 'template' value allows you to specify a custom format for messages
|
||||
; written to syslog. This is similar to how cdr_custom is configured.
|
||||
;
|
||||
; Allowed values: A diaplan style string.
|
||||
; Default value: None, this is required field.
|
||||
;
|
||||
; Note: Because of the way substitution is done, the only meaningful values
|
||||
; available when the record is logged are those available via the CDR()
|
||||
; dialplan function. All other channel variables will be unavailable.
|
||||
|
||||
;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
|
||||
|
||||
; High Resolution Time for billsec and duration fields
|
||||
;template = "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration,f)}","${CDR(billsec,f)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}"
|
||||
;[cdr-simple]
|
||||
|
||||
; Since we don't specify a facility or priority for this logging location, the
|
||||
; records will use the defaults specified in the [general] section.
|
||||
|
||||
;template = "We received a call from ${CDR(src)}"
|
||||
77
asterisk/cdr_tds.conf
Executable file
77
asterisk/cdr_tds.conf
Executable file
@ -0,0 +1,77 @@
|
||||
;
|
||||
; Asterisk Call Detail Records (CDR) - FreeTDS Backend
|
||||
;
|
||||
|
||||
;[global]
|
||||
|
||||
; Connection
|
||||
;
|
||||
; Use the 'connection' keyword to specify one of the instance names from your
|
||||
; 'freetds.conf' file. Note that 'freetds.conf' is not an Asterisk
|
||||
; configuration file, but one specific to the FreeTDS library. See the FreeTDS
|
||||
; documentation on 'freetds.conf' for more information:
|
||||
;
|
||||
; http://www.freetds.org/userguide/freetdsconf.htm
|
||||
;
|
||||
; Accepted values: One of the connections specified in freetds.conf
|
||||
|
||||
;connection=ConnectionFromFreeTDSConf
|
||||
|
||||
; Database Name
|
||||
;
|
||||
; The 'dbname' keyword specifies the database name to use when logging CDRs.
|
||||
;
|
||||
; Accepted values: Any valid database name
|
||||
|
||||
;dbname=AsteriskCDRs
|
||||
|
||||
; Database Table Name
|
||||
;
|
||||
; The 'table' keyword identifies which database table is used to log CDRs.
|
||||
;
|
||||
; Accepted value: Any valid table name
|
||||
; Default value: If not specified, a table named 'cdr' is assumed
|
||||
|
||||
;table=cdr
|
||||
|
||||
; Credentials
|
||||
;
|
||||
; The 'username' and 'password' keywords specify the user credentials that
|
||||
; Asterisk should use when connecting to the database.
|
||||
;
|
||||
; Accepted value: Any valid username and password
|
||||
|
||||
;username=mangUsr
|
||||
;password=
|
||||
|
||||
; Language
|
||||
;
|
||||
; The 'language' keyword changes the language which are used for error and
|
||||
; information messages returned by SQL Server. Each database and user has their
|
||||
; own default value, and this default can be overriden here.
|
||||
;
|
||||
; Accepted value: Any language installed on the target SQL Server.
|
||||
; Default value: us_english
|
||||
|
||||
;language=us_english
|
||||
|
||||
; Character Set
|
||||
;
|
||||
; The 'charset' setting is used to change the character set used when connecting
|
||||
; to the database server. Each database and database user has their own
|
||||
; character set setting, and this default can be overriden here.
|
||||
;
|
||||
; Accepted value: Any valid character set available on the target SQL server.
|
||||
; Default value: iso_1
|
||||
|
||||
;charset=BIG5
|
||||
|
||||
; High Resolution Times
|
||||
;
|
||||
; The 'hrtime' setting is used to store high resolution (sub second) times for
|
||||
; billsec and duration fields.
|
||||
;
|
||||
; Accepted value: true or false
|
||||
; Default value: false
|
||||
|
||||
;hrtime=false
|
||||
117
asterisk/cel.conf
Executable file
117
asterisk/cel.conf
Executable file
@ -0,0 +1,117 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL)
|
||||
;
|
||||
|
||||
; Channel Event Logging is a mechanism to provide fine-grained event information
|
||||
; that can be used to generate billing information. Such event information can
|
||||
; be recorded to various backend modules.
|
||||
;
|
||||
|
||||
[general]
|
||||
|
||||
; CEL Activation
|
||||
;
|
||||
; Use the 'enable' keyword to turn CEL on or off.
|
||||
;
|
||||
; Accepted values: yes and no
|
||||
; Default value: no
|
||||
|
||||
;enable=yes
|
||||
|
||||
; Application Tracking
|
||||
;
|
||||
; Use the 'apps' keyword to specify the list of applications for which you want
|
||||
; to receive CEL events. This is a comma separated list of Asterisk dialplan
|
||||
; applications, such as Dial, Queue, and Park.
|
||||
;
|
||||
; Accepted values: A comma separated list of Asterisk dialplan applications
|
||||
; Default value: none
|
||||
;
|
||||
; Note: You may also use 'all' which will result in CEL events being reported
|
||||
; for all Asterisk applications. This may affect Asterisk's performance
|
||||
; significantly.
|
||||
|
||||
apps=dial,park
|
||||
|
||||
; Event Tracking
|
||||
;
|
||||
; Use the 'events' keyword to specify the list of events which you want to be
|
||||
; raised when they occur. This is a comma separated list of the values in the
|
||||
; table below.
|
||||
;
|
||||
; Accepted values: A comma separated list of one or more of the following:
|
||||
; ALL -- Generate entries on all events
|
||||
; CHAN_START -- The time a channel was created
|
||||
; CHAN_END -- The time a channel was terminated
|
||||
; ANSWER -- The time a channel was answered (ie, phone taken off-hook)
|
||||
; HANGUP -- The time at which a hangup occurred
|
||||
; BRIDGE_ENTER -- The time a channel was connected into a conference room
|
||||
; BRIDGE_EXIT -- The time a channel was removed from a conference room
|
||||
; APP_START -- The time a tracked application was started
|
||||
; APP_END -- the time a tracked application ended
|
||||
; PARK_START -- The time a call was parked
|
||||
; PARK_END -- Unpark event
|
||||
; BLINDTRANSFER -- When a blind transfer is initiated
|
||||
; ATTENDEDTRANSFER -- When an attended transfer is initiated
|
||||
; PICKUP -- This channel picked up the specified channel
|
||||
; FORWARD -- This channel is being forwarded somewhere else
|
||||
; LINKEDID_END -- The last channel with the given linkedid is retired
|
||||
; USER_DEFINED -- Triggered from the dialplan, and has a name given by the
|
||||
; user
|
||||
; LOCAL_OPTIMIZE -- A local channel pair is optimizing away.
|
||||
;
|
||||
; Default value: none
|
||||
; (Track no events)
|
||||
|
||||
events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT
|
||||
|
||||
; Date Format
|
||||
;
|
||||
; Use the 'dateformat' keyword to specify the date format used when CEL events
|
||||
; are raised.
|
||||
;
|
||||
; Accepted values: A strftime format string (see man strftime)
|
||||
;
|
||||
; Example: "%F %T"
|
||||
; -> This gives the date and time in the format "2009-06-23 17:02:35"
|
||||
;
|
||||
; If this option is not specified, the default format is "<seconds>.<microseconds>"
|
||||
; since epoch. The microseconds field will always be 6 digits in length, meaning it
|
||||
; may have leading zeros.
|
||||
;
|
||||
;dateformat = %F %T
|
||||
|
||||
;
|
||||
; Asterisk Manager Interface (AMI) CEL Backend
|
||||
;
|
||||
[manager]
|
||||
|
||||
; AMI Backend Activation
|
||||
;
|
||||
; Use the 'enable' keyword to turn CEL logging to the Asterisk Manager Interface
|
||||
; on or off.
|
||||
;
|
||||
; Accepted values: yes and no
|
||||
; Default value: no
|
||||
;enabled=yes
|
||||
|
||||
; Use 'show_user_defined' to put "USER_DEFINED" in the EventName header,
|
||||
; instead of (by default) just putting the user defined event name there.
|
||||
; When enabled the UserDefType header is added for user defined events to
|
||||
; provide the user defined event name.
|
||||
;
|
||||
;show_user_defined=yes
|
||||
|
||||
;
|
||||
; RADIUS CEL Backend
|
||||
;
|
||||
[radius]
|
||||
;
|
||||
; Log date/time in GMT
|
||||
;usegmtime=yes
|
||||
;
|
||||
; Set this to the location of the radiusclient-ng configuration file
|
||||
; The default is /etc/radiusclient-ng/radiusclient.conf
|
||||
;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
|
||||
radiuscfg => /etc/radcli/radiusclient.conf
|
||||
;
|
||||
21
asterisk/cel_beanstalkd.conf
Executable file
21
asterisk/cel_beanstalkd.conf
Executable file
@ -0,0 +1,21 @@
|
||||
;
|
||||
; Beanstalkd Job Queue Server CEL Backend
|
||||
;
|
||||
[general]
|
||||
|
||||
; Backend Activation
|
||||
;
|
||||
; Use the 'enabled' keyword to turn CEL logging
|
||||
; on or off.
|
||||
;
|
||||
; Accepted values: yes and no
|
||||
; Default value: no
|
||||
;enabled = yes
|
||||
|
||||
;host = 127.0.0.1 ; Specify the remote IP address of the Beanstalkd server
|
||||
;port = 11300 ; Specify the remote PORT of the the Beanstalkd server
|
||||
;tube = asterisk-cel ; Specify the default CDR job queue to use
|
||||
;priority = 99 ; Specify the default job priority for the queue. This parameter is useful when building
|
||||
; platform with multiple Asterisk servers, that are used for different functions. For example,
|
||||
; none billable CDR records can be inserted with a priority of 99, while billable ones be
|
||||
; inserted with a priority of 1
|
||||
35
asterisk/cel_custom.conf
Executable file
35
asterisk/cel_custom.conf
Executable file
@ -0,0 +1,35 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL) - Custom CSV Backend
|
||||
;
|
||||
|
||||
; This is the configuration file for the customizable CSV backend for CEL
|
||||
; logging.
|
||||
;
|
||||
; In order to create custom CSV logs for CEL, uncomment the template below
|
||||
; (Master.csv) and start Asterisk. Once CEL events are generated, a file will
|
||||
; appear in the following location:
|
||||
;
|
||||
; /var/log/asterisk/cel-custom/Master.csv
|
||||
;
|
||||
; (Note that /var/log/asterisk is the default and may differ on your system)
|
||||
;
|
||||
; You can also create more than one template if desired. All logs will appear
|
||||
; in the cel-custom directory under your Asterisk logs directory.
|
||||
;
|
||||
|
||||
;
|
||||
; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
|
||||
; details from the CEL event. There are also a few variables created by this
|
||||
; module that can be used in a mapping:
|
||||
;
|
||||
; eventtype - The name of the CEL event.
|
||||
; eventtime - The timestamp of the CEL event.
|
||||
; eventenum - Like eventtype but is "USER_DEFINED" for a user defined event.
|
||||
; userdeftype - User defined event type name from CELGenUserEvent().
|
||||
; eventextra - Extra data included with this CEL event, typically along with
|
||||
; an event of type USER_DEFINED from CELGenUserEvent().
|
||||
; BRIDGEPEER - Bridged peer channel name at the time of the CEL event.
|
||||
; CHANNEL(peer) could also be used.
|
||||
;
|
||||
[mappings]
|
||||
;Master.csv => ${CSV_QUOTE(${eventtype})},${CSV_QUOTE(${eventtime})},${CSV_QUOTE(${CALLERID(name)})},${CSV_QUOTE(${CALLERID(num)})},${CSV_QUOTE(${CALLERID(ANI)})},${CSV_QUOTE(${CALLERID(RDNIS)})},${CSV_QUOTE(${CALLERID(DNID)})},${CSV_QUOTE(${CHANNEL(exten)})},${CSV_QUOTE(${CHANNEL(context)})},${CSV_QUOTE(${CHANNEL(channame)})},${CSV_QUOTE(${CHANNEL(appname)})},${CSV_QUOTE(${CHANNEL(appdata)})},${CSV_QUOTE(${CHANNEL(amaflags)})},${CSV_QUOTE(${CHANNEL(accountcode)})},${CSV_QUOTE(${CHANNEL(uniqueid)})},${CSV_QUOTE(${CHANNEL(linkedid)})},${CSV_QUOTE(${BRIDGEPEER})},${CSV_QUOTE(${CHANNEL(userfield)})},${CSV_QUOTE(${userdeftype})},${CSV_QUOTE(${eventextra})}
|
||||
108
asterisk/cel_odbc.conf
Executable file
108
asterisk/cel_odbc.conf
Executable file
@ -0,0 +1,108 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL) - Adaptive ODBC Backend
|
||||
;
|
||||
|
||||
; General module options category.
|
||||
[general]
|
||||
; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
|
||||
; instead of (by default) just putting the user defined event name there.
|
||||
;
|
||||
;show_user_defined=yes
|
||||
|
||||
; This configuration defines the connections and tables for which CEL records
|
||||
; may be populated. Each context specifies a different CEL table to be used.
|
||||
;
|
||||
; The columns in the tables should match up word-for-word (case-insensitive) to
|
||||
; the CEL variables set in the dialplan. The natural advantage to this system
|
||||
; is that beyond setting up the configuration file to tell you what tables to
|
||||
; look at, there isn't anything more to do beyond creating the columns for the
|
||||
; fields that you want, and populating the corresponding CEL variables in the
|
||||
; dialplan.
|
||||
;
|
||||
; Please note that after adding columns to the database, it is necessary to
|
||||
; reload this module to get the new column names and types read.
|
||||
;
|
||||
; Warning: if you specify two contexts with exactly the same connection and
|
||||
; table names, you will get duplicate records in that table. So be careful.
|
||||
;
|
||||
; CEL FIELDS:
|
||||
; eventtype
|
||||
; CHANNEL_START = 1
|
||||
; CHANNEL_END = 2
|
||||
; HANGUP = 3
|
||||
; ANSWER = 4
|
||||
; APP_START = 5
|
||||
; APP_END = 6
|
||||
; BRIDGE_START = 7
|
||||
; BRIDGE_END = 8
|
||||
; CONF_START = 9
|
||||
; CONF_END = 10
|
||||
; PARK_START = 11
|
||||
; PARK_END = 12
|
||||
; BLINDTRANSFER = 13
|
||||
; ATTENDEDTRANSFER = 14
|
||||
; TRANSFER = 15
|
||||
; HOOKFLASH = 16
|
||||
; 3WAY_START = 17
|
||||
; 3WAY_END = 18
|
||||
; CONF_ENTER = 19
|
||||
; CONF_EXIT = 20
|
||||
; USER_DEFINED = 21
|
||||
; LINKEDID_END = 22
|
||||
; BRIDGE_UPDATE = 23
|
||||
; PICKUP = 24
|
||||
; FORWARD = 25
|
||||
; eventtime (timeval, includes microseconds)
|
||||
; userdeftype (set only if eventtype == USER_DEFINED)
|
||||
; cid_name
|
||||
; cid_num
|
||||
; cid_ani
|
||||
; cid_rdnis
|
||||
; cid_dnid
|
||||
; exten
|
||||
; context
|
||||
; channame
|
||||
; appname
|
||||
; appdata
|
||||
; accountcode
|
||||
; peeraccount
|
||||
; uniqueid
|
||||
; linkedid
|
||||
; amaflags (an int)
|
||||
; userfield
|
||||
; peer
|
||||
; extra
|
||||
|
||||
; The point of this module is to allow you log whatever you like in terms of the
|
||||
; CEL variables. Do you want to log uniqueid? Then simply ensure that your
|
||||
; table has that column. If you don't want the column, ensure that it does not
|
||||
; exist in the table structure. If you'd like to call uniqueid something else
|
||||
; in your table, simply provide an alias in this file that maps the standard CEL
|
||||
; field name (uniqueid) to whatever column name you like.
|
||||
|
||||
;[first]
|
||||
;connection=mysql1
|
||||
;table=cel
|
||||
|
||||
;[second]
|
||||
;connection=mysql1
|
||||
;table=extracel
|
||||
|
||||
;[third]
|
||||
;connection=sqlserver
|
||||
;table=AsteriskCEL
|
||||
;usegmtime=yes ; defaults to no
|
||||
;allowleapsecond=no ; allow leap second in SQL column for eventtime, default yes.
|
||||
;alias src => source
|
||||
;alias channel => source_channel
|
||||
;alias dst => dest
|
||||
;alias dstchannel => dest_channel
|
||||
|
||||
; Any filter specified MUST match exactly or the event will be discarded
|
||||
;filter accountcode => somename
|
||||
;filter src => 123
|
||||
|
||||
; Additionally, we now support setting static values per column. Reason
|
||||
; for this is to allow different sections to specify different values for
|
||||
; a certain named column, presumably separated by filters.
|
||||
;static "Some Special Value" => identifier_code
|
||||
73
asterisk/cel_pgsql.conf
Executable file
73
asterisk/cel_pgsql.conf
Executable file
@ -0,0 +1,73 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL) - PostgreSQL Backend
|
||||
;
|
||||
|
||||
; Sample Asterisk config file for CEL logging to PostgreSQL
|
||||
;
|
||||
; CEL field names:
|
||||
;
|
||||
; eventtype
|
||||
; CHANNEL_START = 1
|
||||
; CHANNEL_END = 2
|
||||
; HANGUP = 3
|
||||
; ANSWER = 4
|
||||
; APP_START = 5
|
||||
; APP_END = 6
|
||||
; BRIDGE_START = 7
|
||||
; BRIDGE_END = 8
|
||||
; CONF_START = 9
|
||||
; CONF_END = 10
|
||||
; PARK_START = 11
|
||||
; PARK_END = 12
|
||||
; BLINDTRANSFER = 13
|
||||
; ATTENDEDTRANSFER = 14
|
||||
; TRANSFER = 15
|
||||
; HOOKFLASH = 16
|
||||
; 3WAY_START = 17
|
||||
; 3WAY_END = 18
|
||||
; CONF_ENTER = 19
|
||||
; CONF_EXIT = 20
|
||||
; USER_DEFINED = 21
|
||||
; LINKEDID_END = 22
|
||||
; BRIDGE_UPDATE = 23
|
||||
; PICKUP = 24
|
||||
; FORWARD = 25
|
||||
; eventtime (timeval, includes microseconds)
|
||||
; userdeftype (set only if eventtype == USER_DEFINED)
|
||||
; cid_name
|
||||
; cid_num
|
||||
; cid_ani
|
||||
; cid_rdnis
|
||||
; cid_dnid
|
||||
; exten
|
||||
; context
|
||||
; channame
|
||||
; appname
|
||||
; appdata
|
||||
; accountcode
|
||||
; peeraccount
|
||||
; uniqueid
|
||||
; linkedid
|
||||
; amaflags (an int)
|
||||
; userfield
|
||||
; peer
|
||||
; extra
|
||||
|
||||
[global]
|
||||
; Use 'show_user_defined' to put "USER_DEFINED" in the eventtype field,
|
||||
; instead of (by default) just putting the user defined event name there.
|
||||
;
|
||||
;show_user_defined=yes
|
||||
|
||||
; Log date/time in GMT. The default of this option is 'no'.
|
||||
;usegmtime=yes
|
||||
|
||||
;hostname=localhost
|
||||
;port=5432
|
||||
;dbname=asterisk
|
||||
;password=password
|
||||
;user=postgres
|
||||
;table=cel ;SQL table where CEL's will be inserted
|
||||
;schema=public ;Schema where CEL's table is located. Optional parameter.
|
||||
;If schema support is present the default value used will be current_schema().
|
||||
;appname=asterisk ; Postgres application_name support (optional). Whitespace not allowed.
|
||||
25
asterisk/cel_sqlite3_custom.conf
Executable file
25
asterisk/cel_sqlite3_custom.conf
Executable file
@ -0,0 +1,25 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL) - SQLite 3 Backend
|
||||
;
|
||||
|
||||
;
|
||||
; Mappings for sqlite3 config file
|
||||
;
|
||||
; Within a mapping, use the CALLERID() and CHANNEL() functions to retrieve
|
||||
; details from the CEL event. There are also a few variables created by this
|
||||
; module that can be used in a mapping:
|
||||
;
|
||||
; eventtype - The name of the CEL event.
|
||||
; eventtime - The timestamp of the CEL event.
|
||||
; eventenum - Like eventtype but is "USER_DEFINED" for a user defined event.
|
||||
; userdeftype - User defined event type name from CELGenUserEvent().
|
||||
; eventextra - Extra data included with this CEL event, typically along with
|
||||
; an event of type USER_DEFINED from CELGenUserEvent().
|
||||
; BRIDGEPEER - Bridged peer channel name at the time of the CEL event.
|
||||
; CHANNEL(peer) could also be used.
|
||||
;
|
||||
;[master] ; currently, only file "master.db" is supported, with only one table at a time.
|
||||
;table => cel
|
||||
;columns => eventtype, eventtime, cidname, cidnum, cidani, cidrdnis, ciddnid, context, exten, channame, appname, appdata, amaflags, accountcode, uniqueid, userfield, peer, userdeftype, eventextra
|
||||
;values => '${eventtype}','${eventtime}','${CALLERID(name)}','${CALLERID(num)}','${CALLERID(ANI)}','${CALLERID(RDNIS)}','${CALLERID(DNID)}','${CHANNEL(context)}','${CHANNEL(exten)}','${CHANNEL(channame)}','${CHANNEL(appname)}','${CHANNEL(appdata)}','${CHANNEL(amaflags)}','${CHANNEL(accountcode)}','${CHANNEL(uniqueid)}','${CHANNEL(userfield)}','${BRIDGEPEER}','${userdeftype}','${eventextra}'
|
||||
;busy_timeout => 1000
|
||||
69
asterisk/cel_tds.conf
Executable file
69
asterisk/cel_tds.conf
Executable file
@ -0,0 +1,69 @@
|
||||
;
|
||||
; Asterisk Channel Event Logging (CEL) - FreeTDS Backend
|
||||
;
|
||||
|
||||
;[global]
|
||||
|
||||
; Connection
|
||||
;
|
||||
; Use the 'connection' keyword to specify one of the instance names from your
|
||||
; 'freetds.conf' file. Note that 'freetds.conf' is not an Asterisk
|
||||
; configuration file, but one specific to the FreeTDS library. See the FreeTDS
|
||||
; documentation on 'freetds.conf' for more information:
|
||||
;
|
||||
; http://www.freetds.org/userguide/freetdsconf.htm
|
||||
;
|
||||
; Accepted values: One of the connections specified in freetds.conf
|
||||
|
||||
;connection=ConnectionFromFreeTDSConf
|
||||
|
||||
; Database Name
|
||||
;
|
||||
; The 'dbname' keyword specifies the database name to use when logging CEL
|
||||
; records.
|
||||
;
|
||||
; Accepted values: Any valid database name
|
||||
|
||||
;dbname=MalicoHN
|
||||
|
||||
; Database Table Name
|
||||
;
|
||||
; The 'table' keyword identifies which database table is used to log CEL
|
||||
; records.
|
||||
;
|
||||
; Accepted value: Any valid table name
|
||||
; Default value: If not specified, a table named 'cel' is assumed
|
||||
|
||||
;table=cel
|
||||
|
||||
; Credentials
|
||||
;
|
||||
; The 'username' and 'password' keywords specify the user credentials that
|
||||
; Asterisk should use when connecting to the database.
|
||||
;
|
||||
; Accepted value: Any valid username and password
|
||||
|
||||
;username=mangUsr
|
||||
;password=
|
||||
|
||||
; Language
|
||||
;
|
||||
; The 'language' keyword changes the language which are used for error and
|
||||
; information messages returned by SQL Server. Each database and user has their
|
||||
; own default value, and this default can be overriden here.
|
||||
;
|
||||
; Accepted value: Any language installed on the target SQL Server.
|
||||
; Default value: Server default
|
||||
|
||||
;language=us_english
|
||||
|
||||
; Character Set
|
||||
;
|
||||
; The 'charset' setting is used to change the character set used when connecting
|
||||
; to the database server. Each database and database user has their own
|
||||
; character set setting, and this default can be overriden here.
|
||||
;
|
||||
; Accepted value: Any valid character set available on the target server.
|
||||
; Default value: Server setting
|
||||
|
||||
;charset=BIG5
|
||||
1717
asterisk/chan_dahdi.conf
Executable file
1717
asterisk/chan_dahdi.conf
Executable file
File diff suppressed because it is too large
Load Diff
69
asterisk/chan_mobile.conf
Executable file
69
asterisk/chan_mobile.conf
Executable file
@ -0,0 +1,69 @@
|
||||
;
|
||||
; chan_mobile.conf
|
||||
; configuration file for chan_mobile
|
||||
;
|
||||
|
||||
[general]
|
||||
interval=30 ; Number of seconds between trying to connect to devices.
|
||||
|
||||
; The following is a list of adapters we use.
|
||||
; id must be unique and address is the bdaddr of the adapter from hciconfig.
|
||||
; Each adapter may only have one device (headset or phone) connected at a time.
|
||||
; Add an [adapter] entry for each adapter you have.
|
||||
|
||||
[adapter]
|
||||
id=blue
|
||||
address=00:09:DD:60:01:A3
|
||||
;forcemaster=yes ; attempt to force adapter into master mode. default is no.
|
||||
;alignmentdetection=yes ; enable this if you sometimes get 'white noise' on asterisk side of the call
|
||||
; its a bug in the bluetooth adapter firmware, enabling this will compensate for it.
|
||||
; default is no.
|
||||
|
||||
[adapter]
|
||||
id=dlink
|
||||
address=00:80:C8:35:52:78
|
||||
|
||||
; The following is a list of the devices we deal with.
|
||||
; Every device listed below will be available for calls in and out of Asterisk.
|
||||
; Each device needs an adapter=xxxx entry which determines which bluetooth adapter is used.
|
||||
; Use the CLI command 'mobile search' to discover devices.
|
||||
; Use the CLI command 'mobile show devices' to see device status.
|
||||
;
|
||||
; To place a call out through a mobile phone use Dial(Mobile/[device]/NNN.....) or Dial(Mobile/gn/NNN......) in your dialplan.
|
||||
; To call a headset use Dial(Mobile/[device]).
|
||||
|
||||
[LGTU550]
|
||||
address=00:E0:91:7F:46:44 ; the address of the phone
|
||||
port=4 ; the rfcomm port number (from mobile search)
|
||||
context=incoming-mobile ; dialplan context for incoming calls
|
||||
adapter=dlink ; adapter to use
|
||||
group=1 ; this phone is in channel group 1
|
||||
;sms=no ; support SMS, defaults to yes
|
||||
;nocallsetup=yes ; set this only if your phone reports that it supports call progress notification, but does not do it. Motorola L6 for example.
|
||||
|
||||
[blackberry]
|
||||
address=00:60:57:32:7E:B2
|
||||
port=2
|
||||
context=incoming-mobile
|
||||
adapter=dlink
|
||||
group=1
|
||||
;blackberry=yes ; set this if you are using a blackberry device
|
||||
|
||||
[6310i]
|
||||
address=00:60:57:32:7E:B1
|
||||
port=13
|
||||
context=incoming-mobile
|
||||
adapter=dlink
|
||||
group=1 ; this phone is in channel group 1 also.
|
||||
|
||||
[headset]
|
||||
address=00:0B:9E:11:AE:C6
|
||||
port=1
|
||||
type=headset ; This is a headset, not a Phone !
|
||||
adapter=blue
|
||||
|
||||
[headset1]
|
||||
address=00:0B:9E:11:74:A5
|
||||
port=1
|
||||
type=headset
|
||||
adapter=dlink
|
||||
12
asterisk/cli.conf
Executable file
12
asterisk/cli.conf
Executable file
@ -0,0 +1,12 @@
|
||||
;
|
||||
; Asterisk CLI configuration
|
||||
;
|
||||
|
||||
[startup_commands]
|
||||
;
|
||||
; Any commands listed in this section will get automatically executed
|
||||
; when Asterisk starts as a daemon or foreground process (-c).
|
||||
;
|
||||
;sip set debug on = yes
|
||||
;core set verbose 3 = yes
|
||||
;core set debug 1 = yes
|
||||
203
asterisk/cli_aliases.conf
Executable file
203
asterisk/cli_aliases.conf
Executable file
@ -0,0 +1,203 @@
|
||||
;
|
||||
; CLI Aliases configuration
|
||||
;
|
||||
; This module also registers a "cli show aliases" CLI command to list
|
||||
; configured CLI aliases.
|
||||
|
||||
[general]
|
||||
; Here you define what alias templates you want to use. You can also define
|
||||
; multiple templates to use as well. If you do, and there is a conflict, then
|
||||
; the first alias defined will win.
|
||||
;
|
||||
template = friendly ; By default, include friendly aliases
|
||||
;template = asterisk_1dot2 ; Asterisk 1.2 style syntax
|
||||
;template = asterisk_1dot4 ; Asterisk 1.4 style syntax
|
||||
;template = individual_custom ; see [individual_custom] example below which
|
||||
; includes a list of aliases from an external
|
||||
; file
|
||||
|
||||
|
||||
; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
|
||||
; sometimes we run into an issue between being consistant with this format
|
||||
; in the core system, and maintaining system friendliness. In order to get
|
||||
; around this we're providing some useful aliases by default.
|
||||
;
|
||||
[friendly]
|
||||
hangup request=channel request hangup
|
||||
originate=channel originate
|
||||
help=core show help
|
||||
pri intense debug span=pri set debug intense span
|
||||
reload=module reload
|
||||
pjsip reload=module reload res_pjsip.so res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so
|
||||
|
||||
; CLI Alias Templates
|
||||
; -------------------
|
||||
;
|
||||
; You can define several alias templates.
|
||||
; It works with context templates like all other configuration files
|
||||
;
|
||||
;[asterisk](!)
|
||||
; To create an alias you simply set the variable name as the alias and variable
|
||||
; value as the real CLI command you want executed
|
||||
;
|
||||
;die die die=stop now
|
||||
|
||||
;[asterisk_1dot6](asterisk)
|
||||
; Alias for making voicemail reload actually do module reload app_voicemail.so
|
||||
;voicemail reload=module reload app_voicemail.so
|
||||
; This will make the CLI command "mr" behave as though it is "module reload".
|
||||
;mr=module reload
|
||||
;
|
||||
;
|
||||
; In addition, you could also include a flat file of aliases which is loaded by
|
||||
; the [individual_custom] template in the [general] section.
|
||||
;
|
||||
;[individual_custom]
|
||||
;#include "/etc/asterisk/aliases"
|
||||
|
||||
; Implemented CLI Alias Templates
|
||||
; -------------------------------
|
||||
;
|
||||
; Below here we have provided you with some templates, easily allowing you to
|
||||
; utilize previous Asterisk CLI commands with any version of Asterisk. In this
|
||||
; way you will be able to use Asterisk 1.2 and 1.4 style CLI syntax with any
|
||||
; version Asterisk going forward into the future.
|
||||
;
|
||||
; We have also separated out the vanilla syntax into a context template which
|
||||
; allows you to keep your custom changes separate of the standard templates
|
||||
; we have provided you. In this way you can clearly see your custom changes,
|
||||
; and also allowing you to combine various templates as you see fit.
|
||||
;
|
||||
; The naming scheme we have used is recommended, but certainly is not enforced
|
||||
; by Asterisk. If you wish to use the provided templates, simply define the
|
||||
; context name which does not utilize the '_tpl' at the end. For example,
|
||||
; if you would like to use the Asterisk 1.2 style syntax, define in the
|
||||
; [general] section
|
||||
|
||||
[asterisk_1dot2_tpl](!)
|
||||
show channeltypes=core show channeltypes
|
||||
show channeltype=core show channeltype
|
||||
show manager command=manager show command
|
||||
show manager commands=manager show commands
|
||||
show manager connected=manager show connected
|
||||
show manager eventq=manager show eventq
|
||||
rtp no debug=rtp set debug off
|
||||
rtp rtcp debug ip=rtcp debug ip
|
||||
rtp rtcp debug=rtcp debug
|
||||
rtp rtcp no debug=rtcp debug off
|
||||
rtp rtcp stats=rtcp stats
|
||||
rtp rtcp no stats=rtcp stats off
|
||||
stun no debug=stun debug off
|
||||
udptl no debug=udptl debug off
|
||||
show image formats=core show image formats
|
||||
show file formats=core show file formats
|
||||
show applications=core show applications
|
||||
show functions=core show functions
|
||||
show switches=core show switches
|
||||
show hints=core show hints
|
||||
show globals=core show globals
|
||||
show function=core show function
|
||||
show application=core show application
|
||||
set global=core set global
|
||||
show dialplan=dialplan show
|
||||
show codecs=core show codecs
|
||||
show audio codecs=core show audio codecs
|
||||
show video codecs=core show video codecs
|
||||
show image codecs=core show image codecs
|
||||
show codec=core show codec
|
||||
moh classes show=moh show classes
|
||||
moh files show=moh show files
|
||||
agi no debug=agi debug off
|
||||
show agi=agi show
|
||||
dump agihtml=agi dumphtml
|
||||
show features=feature show
|
||||
show indications=indication show
|
||||
answer=console answer
|
||||
hangup=console hangup
|
||||
flash=console flash
|
||||
dial=console dial
|
||||
mute=console mute
|
||||
unmute=console unmute
|
||||
transfer=console transfer
|
||||
send text=console send text
|
||||
autoanswer=console autoanswer
|
||||
oss boost=console boost
|
||||
console=console active
|
||||
save dialplan=dialplan save
|
||||
add extension=dialplan add extension
|
||||
remove extension=dialplan remove extension
|
||||
add ignorepat=dialplan add ignorepat
|
||||
remove ignorepat=dialplan remove ignorepat
|
||||
include context=dialplan add include
|
||||
dont include=dialplan remove include
|
||||
extensions reload=dialplan reload
|
||||
show translation=core show translation
|
||||
convert=file convert
|
||||
show queue=queue show
|
||||
add queue member=queue add member
|
||||
remove queue member=queue remove member
|
||||
ael no debug=ael nodebug
|
||||
sip debug=sip set debug
|
||||
sip no debug=sip set debug off
|
||||
show voicemail users=voicemail show users
|
||||
show voicemail zones=voicemail show zones
|
||||
iax2 trunk debug=iax2 set debug trunk
|
||||
iax2 jb debug=iax2 set debug jb
|
||||
iax2 no debug=iax2 set debug off
|
||||
iax2 no trunk debug=iax2 set debug trunk off
|
||||
iax2 no jb debug=iax2 set debug jb off
|
||||
show agents=agent show
|
||||
show agents online=agent show online
|
||||
show memory allocations=memory show allocations
|
||||
show memory summary=memory show summary
|
||||
show version=core show version
|
||||
show version files=core show file version
|
||||
show profile=core show profile
|
||||
clear profile=core clear profile
|
||||
soft hangup=channel request hangup
|
||||
|
||||
[asterisk_1dot2](asterisk_1dot2_tpl)
|
||||
; add any additional custom commands you want below here, for example:
|
||||
;die quickly=stop now
|
||||
|
||||
[asterisk_1dot4_tpl](!)
|
||||
cdr status=cdr show status
|
||||
rtp debug=rtp set debug on
|
||||
rtcp debug=rtcp set debug on
|
||||
rtcp stats=rtcp set stats on
|
||||
stun debug=stun set debug on
|
||||
udptl debug=udptl set debug on
|
||||
core show globals=dialplan show globals
|
||||
core set global=dialplan set global
|
||||
core set chanvar=dialplan set chanvar
|
||||
agi dumphtml=agi dump html
|
||||
ael debug=ael set debug
|
||||
funcdevstate list=devstate list
|
||||
sip history=sip set history on
|
||||
skinny debug=skinny set debug on
|
||||
mgcp set debug=mgcp set debug on
|
||||
abort shutdown=core abort shutdown
|
||||
stop now=core stop now
|
||||
stop gracefully=core stop gracefully
|
||||
stop when convenient=core stop when convenient
|
||||
restart now=core restart now
|
||||
restart gracefully=core restart gracefully
|
||||
restart when convenient=core restart when convenient
|
||||
soft hangup=channel request hangup
|
||||
|
||||
[asterisk_1dot4](asterisk_1dot4_tpl)
|
||||
; add any additional custom commands you want below here.
|
||||
|
||||
[asterisk_11_tpl](!)
|
||||
jabber list nodes=xmpp list nodes
|
||||
jabber purge nodes=xmpp purge nodes
|
||||
jabber delete node=xmpp delete node
|
||||
jabber create collection=xmpp create collection
|
||||
jabber create leaf=xmpp create leaf
|
||||
jabber set debug=xmpp set debug
|
||||
jabber show connections=xmpp show connections
|
||||
jabber show buddies=xmpp show buddies
|
||||
features reload=module reload features
|
||||
|
||||
[asterisk_11](asterisk_11_tpl)
|
||||
; add any additional custom commands you want below here.
|
||||
82
asterisk/cli_permissions.conf
Executable file
82
asterisk/cli_permissions.conf
Executable file
@ -0,0 +1,82 @@
|
||||
;
|
||||
; CLI permissions configuration example for Asterisk
|
||||
;
|
||||
; All the users that you want to connect with asterisk using
|
||||
; rasterisk, should have write/read access to the
|
||||
; asterisk socket (asterisk.ctl). You could change the permissions
|
||||
; of this file in 'asterisk.conf' config parameter: 'astctlpermissions' (0666)
|
||||
; found on the [files] section.
|
||||
;
|
||||
; general options:
|
||||
;
|
||||
; default_perm = permit | deny
|
||||
; This is the default permissions to apply for a user that
|
||||
; does not has a permissions definided.
|
||||
;
|
||||
; user options:
|
||||
; permit = <command name> | all ; allow the user to run 'command' |
|
||||
; ; allow the user to run 'all' the commands
|
||||
; deny = <command name> | all ; disallow the user to run 'command' |
|
||||
; ; disallow the user to run 'all' commands.
|
||||
;
|
||||
|
||||
[general]
|
||||
|
||||
default_perm=permit ; To leave asterisk working as normal
|
||||
; we should set this parameter to 'permit'
|
||||
;
|
||||
; Follows the per-users permissions configs.
|
||||
;
|
||||
; This list is read in the sequence that is being written, so
|
||||
; In this example the user 'eliel' is allow to run only the following
|
||||
; commands:
|
||||
; sip show peer
|
||||
; core set debug
|
||||
; core set verbose
|
||||
; If the user is not specified, the default_perm option will be apply to
|
||||
; every command.
|
||||
;
|
||||
; Notice that you can also use regular expressions to allow or deny access to a
|
||||
; certain command like: 'core show application D*'. In this example the user will be
|
||||
; allowed to view the documentation for all the applications starting with 'D'.
|
||||
; Another regular expression could be: 'channel originate SIP/[0-9]* extension *'
|
||||
; allowing the user to use 'channel originate' on a sip channel and with the 'extension'
|
||||
; parameter and avoiding the use of the 'application' parameter.
|
||||
;
|
||||
; We can also use the templates syntax:
|
||||
; [supportTemplate](!)
|
||||
; deny=all
|
||||
; permit=sip show ; all commands starting with 'sip show' will be allowed
|
||||
; permit=core show
|
||||
;
|
||||
; You can specify permissions for a local group instead of a user,
|
||||
; just put a '@' and we will know that is a group.
|
||||
; IMPORTANT NOTE: Users permissions overwrite group permissions.
|
||||
;
|
||||
;[@adm]
|
||||
;deny=all
|
||||
;permit=sip
|
||||
;permit=core
|
||||
;
|
||||
;
|
||||
;[eliel]
|
||||
;deny=all
|
||||
;permit=sip show peer
|
||||
;deny=sip show peers
|
||||
;permit=core set
|
||||
;
|
||||
;
|
||||
;User 'tommy' inherits from template 'supportTemplate':
|
||||
; deny=all
|
||||
; permit=sip show
|
||||
; permit=core show
|
||||
;[tommy](supportTemplate)
|
||||
;permit=core set debug
|
||||
;permit=dialplan show
|
||||
;
|
||||
;
|
||||
;[mark]
|
||||
;deny=all
|
||||
;permit=all
|
||||
;
|
||||
;
|
||||
214
asterisk/codecs.conf
Executable file
214
asterisk/codecs.conf
Executable file
@ -0,0 +1,214 @@
|
||||
[speex]
|
||||
; CBR encoding quality [0..10]
|
||||
; used only when vbr = false
|
||||
quality => 3
|
||||
|
||||
; codec complexity [0..10]
|
||||
; tradeoff between cpu/quality
|
||||
complexity => 2
|
||||
|
||||
; perceptual enhancement [true / false]
|
||||
; improves clarity of decoded speech
|
||||
enhancement => true
|
||||
|
||||
; voice activity detection [true / false]
|
||||
; reduces bitrate when no voice detected, used only for CBR
|
||||
; (implicit in VBR/ABR)
|
||||
vad => true
|
||||
|
||||
; variable bit rate [true / false]
|
||||
; uses bit rate proportionate to voice complexity
|
||||
vbr => true
|
||||
|
||||
; available bit rate [bps, 0 = off]
|
||||
; encoding quality modulated to match this target bit rate
|
||||
; not recommended with dtx or pp_vad - may cause bandwidth spikes
|
||||
abr => 0
|
||||
|
||||
; VBR encoding quality [0-10]
|
||||
; floating-point values allowed
|
||||
vbr_quality => 4
|
||||
|
||||
; discontinuous transmission [true / false]
|
||||
; stops transmitting completely when silence is detected
|
||||
; pp_vad is far more effective but more CPU intensive
|
||||
dtx => false
|
||||
|
||||
; preprocessor configuration
|
||||
; these options only affect Speex v1.1.8 or newer
|
||||
|
||||
; enable preprocessor [true / false]
|
||||
; allows dsp functionality below but incurs CPU overhead
|
||||
preprocess => false
|
||||
|
||||
; preproc voice activity detection [true / false]
|
||||
; more advanced equivalent of DTX, based on voice frequencies
|
||||
pp_vad => false
|
||||
|
||||
; preproc automatic gain control [true / false]
|
||||
pp_agc => false
|
||||
pp_agc_level => 8000
|
||||
|
||||
; preproc denoiser [true / false]
|
||||
pp_denoise => false
|
||||
|
||||
; preproc dereverb [true / false]
|
||||
pp_dereverb => false
|
||||
pp_dereverb_decay => 0.4
|
||||
pp_dereverb_level => 0.3
|
||||
|
||||
; experimental bitrate changes depending on RTCP feedback [true / false]
|
||||
experimental_rtcp_feedback => false
|
||||
|
||||
|
||||
[plc]
|
||||
; for all codecs which do not support native PLC
|
||||
; this determines whether to perform generic PLC
|
||||
; there is a minor performance penalty for this.
|
||||
; By default plc is applied only when the 2 codecs
|
||||
; in a channel are different.
|
||||
genericplc => true
|
||||
; Apply generic plc to channels even if the 2 codecs
|
||||
; are the same. This forces transcoding via slin so
|
||||
; the performance impact should be considered.
|
||||
; Ignored if genericplc is not also enabled.
|
||||
genericplc_on_equal_codecs => false
|
||||
|
||||
; Generate custom formats for formats requiring attributes.
|
||||
; After defining the custom format, the name used in defining
|
||||
; the format can be used throughout Asterisk in the format 'allow'
|
||||
; and 'disallow' options.
|
||||
;
|
||||
; Example: silk8 is a predefined custom format in this config file.
|
||||
; Once this config file is loaded, silk8 can be used anywhere a
|
||||
; peer's codec capabilities are defined.
|
||||
;
|
||||
; In sip.conf 'silk8' can be defined as a capability for a peer.
|
||||
; [peer1]
|
||||
; type=peer
|
||||
; host=dynamic
|
||||
; disallow=all
|
||||
; allow=silk8 ;custom codec defined in codecs.conf
|
||||
;
|
||||
; LIMITATIONS
|
||||
; Custom formats can only be defined at startup. Any changes to this
|
||||
; file made after startup will not take into effect until after Asterisk
|
||||
; is restarted.
|
||||
;
|
||||
|
||||
; Default Custom SILK format definitions, only one custom SILK format per
|
||||
; sample rate is allowed.
|
||||
[silk8]
|
||||
type=silk
|
||||
samprate=8000
|
||||
fec=true ; turn on or off encoding with forward error correction.
|
||||
; On recommended, off by default.
|
||||
packetloss_percentage=10 ; Estimated packet loss percentage in uplink direction. This
|
||||
; affects how much redundancy is built in when using fec.
|
||||
; The higher the percentage, the larger amount of bandwidth is
|
||||
; used. Default is 0%, 10% is recommended when fec is in use.
|
||||
|
||||
maxbitrate=10000 ; Use the table below to make sure a useful bitrate is choosen
|
||||
; for maxbitrate. If not set or value is not within the bounds
|
||||
; of the encoder, a default value is chosen.
|
||||
;
|
||||
; sample rate | bitrate range
|
||||
; 8khz | 5000 - 20000 bps
|
||||
; 12khz | 7000 - 25000 bps
|
||||
; 16khz | 8000 - 30000 bps
|
||||
; 24khz | 20000- 40000 bps
|
||||
;
|
||||
;dtx=true ; Encode using discontinuous transmission mode or not. Turning this
|
||||
; on will save bandwidth during periods of silence at the cost of
|
||||
; increased computational complexity. Off by default.
|
||||
|
||||
[silk12]
|
||||
type=silk
|
||||
samprate=12000
|
||||
maxbitrate=12000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
[silk16]
|
||||
type=silk
|
||||
samprate=16000
|
||||
maxbitrate=20000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
[silk24]
|
||||
type=silk
|
||||
samprate=24000
|
||||
maxbitrate=30000
|
||||
fec=true
|
||||
packetloss_percentage=10;
|
||||
|
||||
|
||||
; Default custom CELT codec definitions. Only one custom CELT definition is allowed
|
||||
; per a sample rate.
|
||||
;[celt44]
|
||||
;type=celt
|
||||
;samprate=44100 ; The samplerate in hz. This option is required.
|
||||
;framesize=480 ; The framesize option represents the duration of each frame in samples.
|
||||
; This must be a factor of 2. This option is only advertised in an SDP
|
||||
; when it is set. Otherwise a default of framesize of 480 is assumed
|
||||
; internally
|
||||
|
||||
;[celt48]
|
||||
;type=celt
|
||||
;samprate=48000
|
||||
|
||||
;[celt32]
|
||||
;type=celt
|
||||
;samprate=32000
|
||||
|
||||
;============================ OPUS Section Options ============================
|
||||
;
|
||||
; NOTE: Accurate documentation corresponding to your downloaded version of
|
||||
; codec_opus is available from Asterisk's CLI:
|
||||
;
|
||||
; *CLI> config show help codec_opus opus
|
||||
;
|
||||
;[opus]
|
||||
;type= ; Must be of type "opus" (default: "")
|
||||
;packet_loss= ; Encoder's packet loss percentage. Can be any number between 0
|
||||
; and 100, inclusive. A higher value results in more loss
|
||||
; resistance. (default: 0)
|
||||
;complexity= ; Encoder's computational complexity. Can be any number between 0
|
||||
; and 10, inclusive. Note, 10 equals the highest complexity.
|
||||
; (default: 10)
|
||||
;max_bandwidth= ; Encoder's maximum bandwidth allowed. Sets an upper bandwidth
|
||||
; bound on the encoder. Can be any of the following: narrow,
|
||||
; medium, wide, super_wide, full. (default: full)
|
||||
;signal= ; Encoder's signal type. Aids in mode selection on the encoder: Can
|
||||
; be any of the following: auto, voice, music. (default: auto)
|
||||
;application= ; Encoder's application type. Can be any of the following: voip,
|
||||
; audio, low_delay. (default: voip)
|
||||
;max_playback_rate= ; Override the maximum playback rate in the offer's SDP.
|
||||
; Any value between 8000 and 48000 (inclusive) is valid,
|
||||
; however typically it should match one of the usual opus
|
||||
; bandwidths. (default: 48000)
|
||||
;bitrate= ; Override the maximum average bitrate in the offer's SDP. Any value
|
||||
; between 500 and 512000 is valid. The following values are also
|
||||
; allowed: auto, max. (default: auto)
|
||||
;cbr= ; Override the constant bit rate parameter in the offer's SDP. A value of
|
||||
; 0/false/no represents a variable bit rate whereas 1/true/yes represents
|
||||
; a constant bit rate. (default: no)
|
||||
;fec= ; Override the use inband fec parameter in the offer's SDP. A value of
|
||||
; 0/false/no represents disabled whereas 1/true/yes represents enabled.
|
||||
; (default: yes)
|
||||
;dtx= ; Override the use dtx parameter in the offer's SDP. A value of 0/false/no
|
||||
; represents disabled whereas 1/true/yes represents enabled. (default: no)
|
||||
|
||||
;=============================== OPUS Examples ================================
|
||||
;
|
||||
;[opus]
|
||||
;type=opus
|
||||
;max_playback_rate=8000 ; Limit the maximum playback rate on the encoder
|
||||
;fec=no ; No inband fec
|
||||
|
||||
;[myopus]
|
||||
;type=opus
|
||||
;max_bandwidth=wide ; Maximum encoded bandwidth set to wide band (0-8000 Hz
|
||||
; ; audio bandwidth at 16Khz sample rate)
|
||||
;cbr=yes ; Negotiate a constant bit rate
|
||||
450
asterisk/confbridge.conf
Executable file
450
asterisk/confbridge.conf
Executable file
@ -0,0 +1,450 @@
|
||||
[general]
|
||||
; The general section of this config
|
||||
; is not currently used, but reserved
|
||||
; for future use.
|
||||
|
||||
;
|
||||
; --- Default Information ---
|
||||
; The default_user and default_bridge sections are applied
|
||||
; automatically to all ConfBridge instances invoked without
|
||||
; a user, or bridge argument. No menu is applied by default.
|
||||
;
|
||||
; Note that while properties of the default_user or default_bridge
|
||||
; profile can be overridden, if removed, they will be automatically
|
||||
; added and made available to the dialplan upon module load.
|
||||
;
|
||||
|
||||
; --- ConfBridge User Profile Options ---
|
||||
[default_user]
|
||||
type=user
|
||||
;admin=yes ; Sets if the user is an admin or not. Off by default.
|
||||
|
||||
;send_events=no ; If events are enabled for this bridge and this option is
|
||||
; set, users will receive events like join, leave, talking,
|
||||
; etc. via text messages. For users accessing the bridge
|
||||
; via chan_pjsip, this means in-dialog MESSAGE messages.
|
||||
; This is most useful for WebRTC participants where the
|
||||
; browser application can use the messages to alter the user
|
||||
; interface.
|
||||
;echo_events=yes ; If events are enabled for this user and this option is set,
|
||||
; the user will receive events they trigger, talking, mute, etc.
|
||||
; If not set, they will not receive their own events.
|
||||
|
||||
;marked=yes ; Sets if this is a marked user or not. Off by default.
|
||||
;startmuted=yes; Sets if all users should start out muted. Off by default
|
||||
;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only
|
||||
; one person is in the conference or when the
|
||||
; the user is waiting on a marked user to enter
|
||||
; the conference. Off by default.
|
||||
;music_on_hold_class=default ; The MOH class to use for this user.
|
||||
;quiet=yes ; When enabled enter/leave prompts and user intros are not played.
|
||||
; There are some prompts, such as the prompt to enter a PIN number,
|
||||
; that must be played regardless of what this option is set to.
|
||||
; Off by default
|
||||
;announce_user_count=yes ; Sets if the number of users should be announced to the
|
||||
; caller. Off by default.
|
||||
;announce_user_count_all=yes ; Sets if the number of users should be announced to
|
||||
; all the other users in the conference when someone joins.
|
||||
; This option can be either set to 'yes' or a number.
|
||||
; When set to a number, the announcement will only occur
|
||||
; once the user count is above the specified number.
|
||||
;announce_only_user=yes ; Sets if the only user announcement should be played
|
||||
; when a channel enters a empty conference. On by default.
|
||||
;wait_marked=yes ; Sets if the user must wait for a marked user to enter before
|
||||
; joining the conference. Off by default.
|
||||
;end_marked=yes ; This option will kick every user with this option set in their
|
||||
; user profile after the last Marked user exists the conference.
|
||||
|
||||
;dsp_drop_silence=yes ; This option drops what Asterisk detects as silence from
|
||||
; entering into the bridge. Enabling this option will drastically
|
||||
; improve performance and help remove the buildup of background
|
||||
; noise from the conference. Highly recommended for large conferences
|
||||
; due to its performance enhancements.
|
||||
|
||||
;dsp_talking_threshold=128 ; Average magnitude threshold to determine talking.
|
||||
;
|
||||
; The minimum average magnitude per sample in a frame for the
|
||||
; DSP to consider talking/noise present. A value below this
|
||||
; level is considered silence. This value affects several
|
||||
; operations and should not be changed unless the impact on
|
||||
; call quality is fully understood.
|
||||
;
|
||||
; What this value affects internally:
|
||||
;
|
||||
; 1. Audio is only mixed out of a user's incoming audio
|
||||
; stream if talking is detected. If this value is set too
|
||||
; high the user will hear himself talking.
|
||||
;
|
||||
; 2. When talk detection AMI events are enabled, this value
|
||||
; determines when talking has begun which results in an
|
||||
; AMI event to fire. If this value is set too low AMI
|
||||
; events may be falsely triggered by variants in room
|
||||
; noise.
|
||||
;
|
||||
; 3. The 'drop_silence' option depends on this value to
|
||||
; determine when the user's audio should be mixed into the
|
||||
; bridge after periods of silence. If this value is too
|
||||
; high the user's speech will get discarded as they will
|
||||
; be considered silent.
|
||||
;
|
||||
; Valid values are 1 through 2^15.
|
||||
; By default this value is 160.
|
||||
|
||||
;dsp_silence_threshold=2000 ; The number of milliseconds of silence necessary to declare
|
||||
; talking stopped.
|
||||
;
|
||||
; The time in milliseconds of sound falling below the
|
||||
; 'dsp_talking_threshold' option when a user is considered to
|
||||
; stop talking. This value affects several operations and
|
||||
; should not be changed unless the impact on call quality is
|
||||
; fully understood.
|
||||
;
|
||||
; What this value affects internally:
|
||||
;
|
||||
; 1. When talk detection AMI events are enabled, this value
|
||||
; determines when the user has stopped talking after a
|
||||
; period of talking. If this value is set too low AMI
|
||||
; events indicating the user has stopped talking may get
|
||||
; falsely sent out when the user briefly pauses during mid
|
||||
; sentence.
|
||||
;
|
||||
; 2. The 'drop_silence' option depends on this value to
|
||||
; determine when the user's audio should begin to be
|
||||
; dropped from the conference bridge after the user stops
|
||||
; talking. If this value is set too low the user's audio
|
||||
; stream may sound choppy to the other participants. This
|
||||
; is caused by the user transitioning constantly from
|
||||
; silence to talking during mid sentence.
|
||||
;
|
||||
; The best way to approach this option is to set it slightly
|
||||
; above the maximum amount of milliseconds of silence a user
|
||||
; may generate during natural speech.
|
||||
;
|
||||
; Valid values are 1 through 2^31.
|
||||
; By default this value is 2500ms.
|
||||
|
||||
;talk_detection_events=yes ; This option sets whether or not notifications of when a user
|
||||
; begins and ends talking should be sent out as events over AMI.
|
||||
; By default this option is off.
|
||||
|
||||
;denoise=yes ; Sets whether or not a denoise filter should be applied
|
||||
; to the audio before mixing or not. Off by default. Requires
|
||||
; func_speex to be built and installed. Do not confuse this option
|
||||
; with drop_silence. Denoise is useful if there is a lot of background
|
||||
; noise for a user as it attempts to remove the noise while preserving
|
||||
; the speech. This option does NOT remove silence from being mixed into
|
||||
; the conference and does come at the cost of a slight performance hit.
|
||||
|
||||
;jitterbuffer=yes ; Enabling this option places a jitterbuffer on the user's audio stream
|
||||
; before audio mixing is performed. This is highly recommended but will
|
||||
; add a slight delay to the audio. This option is using the JITTERBUFFER
|
||||
; dialplan function's default adaptive jitterbuffer. For a more fine tuned
|
||||
; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
|
||||
; on the user before entering the ConfBridge application.
|
||||
|
||||
;pin=1234 ; Sets if this user must enter a PIN number before entering
|
||||
; the conference. The PIN will be prompted for.
|
||||
;announce_join_leave=yes ; When enabled, this option will prompt the user for a
|
||||
; name when entering the conference. After the name is
|
||||
; recorded, it will be played as the user enters and exists
|
||||
; the conference. This option is off by default.
|
||||
;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
|
||||
; will be prompted to review their recording before
|
||||
; entering the conference. During this phase, the recording
|
||||
; may be listened to, re-recorded, or accepted as is. This
|
||||
; option is off by default.
|
||||
;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
|
||||
; This option is off by default.
|
||||
;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
|
||||
|
||||
;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
|
||||
; may stay in the conference before being automatically ejected. When the user
|
||||
; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
|
||||
; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
|
||||
; Default: 0
|
||||
;text_messaging=yes ; When set to yes text messages will be sent to this user. Text messages
|
||||
; may occur as a result of events or can be received from other participants.
|
||||
; When set to no text messages will not be sent to this user.
|
||||
|
||||
; --- ConfBridge Bridge Profile Options ---
|
||||
[default_bridge]
|
||||
type=bridge
|
||||
;max_members=50 ; This option limits the number of participants for a single
|
||||
; conference to a specific number. By default conferences
|
||||
; have no participant limit. After the limit is reached, the
|
||||
; conference will be locked until someone leaves. Note however
|
||||
; that an Admin user will always be alowed to join the conference
|
||||
; regardless if this limit is reached or not.
|
||||
|
||||
;record_conference=yes ; Records the conference call starting when the first user
|
||||
; enters the room, and ending when the last user exits the room.
|
||||
; The default recorded filename is
|
||||
; 'confbridge-<name of conference bridge>-<start time>.wav
|
||||
; and the default format is 8khz slinear. This file will be
|
||||
; located in the configured monitoring directory in asterisk.conf.
|
||||
|
||||
;record_file=</path/to/file> ; When record_conference is set to yes, the specific name of the
|
||||
; record file can be set using this option. Note that since multiple
|
||||
; conferences may use the same bridge profile, this may cause issues
|
||||
; depending on the configuration. It is recommended to only use this
|
||||
; option dynamically with the CONFBRIDGE() dialplan function. This
|
||||
; allows the record name to be specified and a unique name to be chosen.
|
||||
; By default, the record_file is stored in Asterisk's spool/monitor directory
|
||||
; with a unique filename starting with the 'confbridge' prefix.
|
||||
|
||||
;record_file_append=yes ; Append record file when starting/stopping on same conference recording.
|
||||
;record_file_timestamp=yes ; Append the start time to the record file name.
|
||||
|
||||
;record_options= ; Pass additional options to MixMonitor.
|
||||
;record_command=</path/to/command> ; Command to execute when recording finishes.
|
||||
|
||||
;internal_sample_rate=auto ; Sets the internal native sample rate the
|
||||
; conference is mixed at. This is set to automatically
|
||||
; adjust the sample rate to the best quality by default.
|
||||
; Other values can be anything from 8000-192000. If a
|
||||
; sample rate is set that Asterisk does not support, the
|
||||
; closest sample rate Asterisk does support to the one requested
|
||||
; will be used.
|
||||
|
||||
;maximum_sample_rate=none ; Sets the maximum sample rate the conference
|
||||
; is mixed at. This is set to no maximum by default.
|
||||
; Values can be anything from 8000-192000.
|
||||
|
||||
;mixing_interval=40 ; Sets the internal mixing interval in milliseconds for the bridge. This
|
||||
; number reflects how tight or loose the mixing will be for the conference.
|
||||
; In order to improve performance a larger mixing interval such as 40ms may
|
||||
; be chosen. Using a larger mixing interval comes at the cost of introducing
|
||||
; larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
|
||||
; or 80. By default 20ms is used.
|
||||
|
||||
;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
|
||||
; Note that participants wanting to view and be the source of a video feed
|
||||
; _MUST_ be sharing the same video codec. Also, using video in conjunction with
|
||||
; with the jitterbuffer currently results in the audio being slightly out of sync
|
||||
; with the video. This is a result of the jitterbuffer only working on the audio
|
||||
; stream. It is recommended to disable the jitterbuffer when video is used.
|
||||
;
|
||||
; --- MODES ---
|
||||
; none: No video sources are set by default in the conference. It is still
|
||||
; possible for a user to be set as a video source via AMI or DTMF action
|
||||
; at any time.
|
||||
;
|
||||
; follow_talker: The video feed will follow whoever is talking and providing video.
|
||||
;
|
||||
; last_marked: The last marked user to join the conference with video capabilities
|
||||
; will be the single source of video distributed to all participants.
|
||||
; If multiple marked users are capable of video, the last one to join
|
||||
; is always the source, when that user leaves it goes to the one who
|
||||
; joined before them.
|
||||
;
|
||||
; first_marked: The first marked user to join the conference with video capabilities
|
||||
; is the single source of video distribution among all participants. If
|
||||
; that user leaves, the marked user to join after them becomes the source.
|
||||
;
|
||||
; sfu: Selective Forwarding Unit - Sets multi-stream operation
|
||||
; for a multi-party video conference.
|
||||
|
||||
;language=en ; Set the language used for announcements to the conference.
|
||||
; Default is en (English).
|
||||
|
||||
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
|
||||
;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
|
||||
; update request. Default is 2000. A video update request is a request for a full video
|
||||
; intra-frame. Clients can request this if they require a full frame in order to decode
|
||||
; the video stream. Since a full frame can be large limiting how often they occur can
|
||||
; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
|
||||
; channel to receive the video stream.
|
||||
;remb_send_interval=1000 ; Interval (in milliseconds) at which a combined REMB frame will be sent to sources of video.
|
||||
; A REMB frame contains receiver estimated maximum bitrate information. By creating a combined
|
||||
; frame and sending it to the sources of video the sender can be influenced on what bitrate
|
||||
; they choose allowing a better experience for the receivers. This defaults to 0, or disabled.
|
||||
;remb_behavior=average ; How the combined REMB report for an SFU video bridge is constructed. If set to "average" then
|
||||
; the estimated maximum bitrate of each receiver is used to construct an average bitrate. If
|
||||
; set to "lowest" the lowest maximum bitrate is forwarded to the sender. If set to "highest"
|
||||
; the highest maximum bitrate is forwarded to the sender. If set to "average_all" a single average
|
||||
; is generated from every receiver and the same value is sent to every sender. If set to
|
||||
; "lowest_all" the lowest maximum bitrate of all receivers is sent to every sender. If set to
|
||||
; "highest_all" the highest maximum bitrate of all receivers is sent to every sender.
|
||||
; When set to "force", the value set in remb_estimated_bitrate is sent to every sender.
|
||||
; This defaults to "average".
|
||||
;remb_estimated_bitrate=0 ; When remb_behavior is set to 'force', this options sets the estimated bitrate
|
||||
; (in bits per second) sent to each participant in REMB reports.
|
||||
|
||||
;enable_events=no ; If enabled, recipients who joined the bridge via a channel driver
|
||||
; that supports Enhanced Messaging (currently only chan_pjsip) will
|
||||
; receive in-dialog messages containing a JSON body describing the
|
||||
; event. The Content-Type header will be
|
||||
; "text/x-ast-confbridge-event".
|
||||
; This feature must also be enabled in user profiles.
|
||||
|
||||
; All sounds in the conference are customizable using the bridge profile options below.
|
||||
; Simply state the option followed by the filename or full path of the filename after
|
||||
; the option. Example: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
|
||||
; sound file found in the sounds directory when announcing someone's name is joining the
|
||||
; conference.
|
||||
|
||||
;sound_join ; The sound played to everyone when someone enters the conference.
|
||||
;sound_leave ; The sound played to everyone when someone leaves the conference.
|
||||
;sound_has_joined ; The sound played before announcing someone's name has
|
||||
; joined the conference. This is used for user intros.
|
||||
; Example "_____ has joined the conference"
|
||||
;sound_has_left ; The sound played when announcing someone's name has
|
||||
; left the conference. This is used for user intros.
|
||||
; Example "_____ has left the conference"
|
||||
;sound_kicked ; The sound played to a user who has been kicked from the conference.
|
||||
;sound_muted ; The sound played when the mute option is toggled on using DTMF menu.
|
||||
;sound_unmuted ; The sound played when the mute option is toggled off using DTMF menu.
|
||||
;sound_only_person ; The sound played when the user is the only person in the conference.
|
||||
;sound_only_one ; The sound played to a user when there is only one other
|
||||
; person is in the conference.
|
||||
;sound_there_are ; The sound played when announcing how many users there
|
||||
; are in a conference.
|
||||
;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
|
||||
; when announcing how many users there are in the conference.
|
||||
; The sounds are stringed together like this.
|
||||
; "sound_there_are" <number of participants> "sound_other_in_party"
|
||||
;sound_place_into_conference ; The sound played when someone is placed into the conference
|
||||
; after waiting for a marked user. This sound is now deprecated
|
||||
; since it was only ever used improperly and correcting that bug
|
||||
; made it completely unused.
|
||||
;sound_wait_for_leader ; The sound played when a user is placed into a conference that
|
||||
; can not start until a marked user enters.
|
||||
;sound_leader_has_left ; The sound played when the last marked user leaves the conference.
|
||||
;sound_get_pin ; The sound played when prompting for a conference pin number.
|
||||
;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
|
||||
;sound_locked ; The sound played to a user trying to join a locked conference.
|
||||
;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
|
||||
;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
|
||||
;sound_error_menu ; The sound played when an invalid menu option is entered.
|
||||
;sound_begin ; The sound played to the conference when the first marked user enters the conference.
|
||||
;sound_binaural_on ; The sound played when binaural audio is turned on
|
||||
;sound_binaural_off ; The sound played when binaural audio is turned off
|
||||
|
||||
; --- ConfBridge Menu Options ---
|
||||
; The ConfBridge application also has the ability to
|
||||
; apply custom DTMF menus to each channel using the
|
||||
; application. Like the User and Bridge profiles
|
||||
; a menu is passed in to ConfBridge as an argument in
|
||||
; the dialplan.
|
||||
;
|
||||
; Below is a list of menu actions that can be assigned
|
||||
; to a DTMF sequence.
|
||||
;
|
||||
; To have the first DTMF digit in a sequence be the '#' character, you need to
|
||||
; escape it. If it is not escaped then normal config file processing will
|
||||
; think it is a directive like #include. For example:
|
||||
; \#1=toggle_mute ; Pressing #1 will toggle the mute setting.
|
||||
;
|
||||
; A single DTMF sequence can have multiple actions associated with it. This is
|
||||
; accomplished by stringing the actions together and using a ',' as the delimiter.
|
||||
; Example: Both listening and talking volume is reset when '5' is pressed.
|
||||
; 5=reset_talking_volume, reset_listening_volume
|
||||
;
|
||||
; playback(<name of audio file>&<name of audio file>)
|
||||
; Playback will play back an audio file to a channel
|
||||
; and then immediately return to the conference.
|
||||
; This file can not be interupted by DTMF.
|
||||
; Mutliple files can be chained together using the
|
||||
; '&' character.
|
||||
; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
|
||||
; playback_and_continue will
|
||||
; play back a prompt while continuing to
|
||||
; collect the dtmf sequence. This is useful
|
||||
; when using a menu prompt that describes all
|
||||
; the menu options. Note however that any DTMF
|
||||
; during this action will terminate the prompts
|
||||
; playback. Prompt files can be chained together
|
||||
; using the '&' character as a delimiter.
|
||||
; toggle_mute ; Toggle turning on and off mute. Mute will make the user silent
|
||||
; to everyone else, but the user will still be able to listen in.
|
||||
; toggle_binaural ; Toggle on or off binaural audio processing.
|
||||
|
||||
; no_op ; This action does nothing (No Operation). Its only real purpose exists for
|
||||
; being able to reserve a sequence in the config as a menu exit sequence.
|
||||
; decrease_listening_volume ; Decreases the channel's listening volume.
|
||||
; increase_listening_volume ; Increases the channel's listening volume.
|
||||
; reset_listening_volume ; Reset channel's listening volume to default level.
|
||||
|
||||
; decrease_talking_volume ; Decreases the channel's talking volume.
|
||||
; increase_talking_volume ; Icreases the channel's talking volume.
|
||||
; reset_talking_volume ; Reset channel's talking volume to default level.
|
||||
;
|
||||
; dialplan_exec(context,exten,priority) ; The dialplan_exec action allows a user
|
||||
; to escape from the conference and execute
|
||||
; commands in the dialplan. Once the dialplan
|
||||
; exits the user will be put back into the
|
||||
; conference. The possibilities are endless!
|
||||
; leave_conference ; This action allows a user to exit the conference and continue
|
||||
; execution in the dialplan.
|
||||
;
|
||||
; admin_kick_last ; This action allows an Admin to kick the last participant from the
|
||||
; conference. This action will only work for admins which allows
|
||||
; a single menu to be used for both users and admins.
|
||||
;
|
||||
; admin_toggle_conference_lock ; This action allows an Admin to toggle locking and
|
||||
; unlocking the conference. Non admins can not use
|
||||
; this action even if it is in their menu.
|
||||
|
||||
; set_as_single_video_src ; This action allows any user to set themselves as the
|
||||
; single video source distributed to all participants.
|
||||
; This will make the video feed stick to them regardless
|
||||
; of what the video_mode is set to.
|
||||
|
||||
; release_as_single_video_src ; This action allows a user to release themselves as
|
||||
; the video source. If video_mode is not set to "none"
|
||||
; this action will result in the conference returning to
|
||||
; whatever video mode the bridge profile is using.
|
||||
;
|
||||
; Note that this action will have no effect if the user
|
||||
; is not currently the video source. Also, the user is
|
||||
; not guaranteed by using this action that they will not
|
||||
; become the video source again. The bridge will return
|
||||
; to whatever operation the video_mode option is set to
|
||||
; upon release of the video src.
|
||||
|
||||
; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
|
||||
; state for all non-admins within a conference.
|
||||
; Subsequent non-admins joining a muted conference will
|
||||
; start muted. All admin users are unaffected by this
|
||||
; option. Note that all users, regardless of their admin
|
||||
; status, are notified that the conference is muted when
|
||||
; the state is toggled.
|
||||
|
||||
; participant_count ; This action plays back the number of participants currently
|
||||
; in a conference
|
||||
|
||||
[sample_user_menu]
|
||||
type=menu
|
||||
*=playback_and_continue(conf-usermenu)
|
||||
*1=toggle_mute
|
||||
1=toggle_mute
|
||||
*4=decrease_listening_volume
|
||||
4=decrease_listening_volume
|
||||
*6=increase_listening_volume
|
||||
6=increase_listening_volume
|
||||
*7=decrease_talking_volume
|
||||
7=decrease_talking_volume
|
||||
*8=leave_conference
|
||||
8=leave_conference
|
||||
*9=increase_talking_volume
|
||||
9=increase_talking_volume
|
||||
|
||||
[sample_admin_menu]
|
||||
type=menu
|
||||
*=playback_and_continue(conf-adminmenu)
|
||||
*1=toggle_mute
|
||||
1=toggle_mute
|
||||
*2=admin_toggle_conference_lock ; only applied to admin users
|
||||
2=admin_toggle_conference_lock ; only applied to admin users
|
||||
*3=admin_kick_last ; only applied to admin users
|
||||
3=admin_kick_last ; only applied to admin users
|
||||
*4=decrease_listening_volume
|
||||
4=decrease_listening_volume
|
||||
*6=increase_listening_volume
|
||||
6=increase_listening_volume
|
||||
*7=decrease_talking_volume
|
||||
7=decrease_talking_volume
|
||||
*8=no_op
|
||||
8=no_op
|
||||
*9=increase_talking_volume
|
||||
9=increase_talking_volume
|
||||
46
asterisk/config_test.conf
Executable file
46
asterisk/config_test.conf
Executable file
@ -0,0 +1,46 @@
|
||||
; Config to test config parsing
|
||||
; global and item have values that differ from defaults
|
||||
; global_defaults and item_defualts are to show all defaults are set
|
||||
; there should be an option for every default type, and a custom type
|
||||
|
||||
[global]
|
||||
intopt=-1
|
||||
uintopt=1
|
||||
timelenopt1=1ms
|
||||
timelenopt2=1s
|
||||
timelenopt3=1m
|
||||
timelenopt4=1h
|
||||
doubleopt=0.1
|
||||
sockaddropt=1.2.3.4:1234
|
||||
boolopt=true
|
||||
boolflag1=true
|
||||
boolflag2=false
|
||||
boolflag3=true
|
||||
deny=0.0.0.0/0
|
||||
permit=1.2.3.4/32
|
||||
codecopt=!all,ulaw,g729
|
||||
stropt=test
|
||||
customopt=yes
|
||||
|
||||
[global_defaults]
|
||||
|
||||
[item]
|
||||
intopt=-1
|
||||
uintopt=1
|
||||
timelenopt1=1
|
||||
timelenopt2=1
|
||||
timelenopt3=1
|
||||
timelenopt4=1
|
||||
doubleopt=0.1
|
||||
sockaddropt=1.2.3.4:1234
|
||||
boolopt=true
|
||||
boolflag1=true
|
||||
boolflag2=false
|
||||
boolflag3=true
|
||||
acldenyopt=0.0.0.0/0
|
||||
aclpermitopt=1.2.3.4/32
|
||||
codecopt=!all,ulaw,g729
|
||||
stropt=test
|
||||
customopt=yes
|
||||
|
||||
[item_defaults]
|
||||
97
asterisk/console.conf
Executable file
97
asterisk/console.conf
Executable file
@ -0,0 +1,97 @@
|
||||
;
|
||||
; Configuration for chan_console, a cross-platform console channel driver.
|
||||
;
|
||||
|
||||
[general]
|
||||
|
||||
; Set this option to "yes" to enable automatically answering calls on the
|
||||
; console. This is very useful if the console is used as an intercom.
|
||||
; The default value is "no".
|
||||
;
|
||||
;autoanswer = no
|
||||
|
||||
; Set the default context to use for outgoing calls. This can be overridden by
|
||||
; dialing some extension@context, unless the overridecontext option is enabled.
|
||||
; The default is "default".
|
||||
;
|
||||
;context = default
|
||||
|
||||
; Set the default extension to use for outgoing calls. The default is "s".
|
||||
;
|
||||
;extension = s
|
||||
|
||||
; Set the default CallerID for created channels.
|
||||
;
|
||||
;callerid = MyName Here <(256) 428-6000>
|
||||
|
||||
; Set the default language for created channels.
|
||||
;
|
||||
;language = en
|
||||
|
||||
; If you set overridecontext to 'yes', then the whole dial string
|
||||
; will be interpreted as an extension, which is extremely useful
|
||||
; to dial SIP, IAX and other extensions which use the '@' character.
|
||||
; The default is "no".
|
||||
;
|
||||
;overridecontext = no ; if 'no', the last @ will start the context
|
||||
; if 'yes' the whole string is an extension.
|
||||
|
||||
|
||||
; Default Music on Hold class to use when this channel is placed on hold in
|
||||
; the case that the music class is not set on the channel with
|
||||
; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
|
||||
; putting this one on hold did not suggest a class to use.
|
||||
;
|
||||
;mohinterpret=default
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; Console channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The Console channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive Console side will always
|
||||
; be used if the sending side can create jitter.
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a Console
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
|
||||
;
|
||||
; Any configuration context defined beyond the [general] section configures
|
||||
; specific devices for use.
|
||||
;
|
||||
|
||||
[default]
|
||||
input_device = default ; When configuring an input device and output device,
|
||||
output_device = default ; use the name that you see when you run the "console
|
||||
; list available" CLI command. If you say "default", the
|
||||
; system default input and output devices will be used.
|
||||
autoanswer = no
|
||||
context = default
|
||||
extension = s
|
||||
callerid = MyName Here <(256) 428-6000>
|
||||
language = en
|
||||
overridecontext = no
|
||||
mohinterpret = default
|
||||
active = yes ; This option should only be set for one console.
|
||||
; It means that it is the active console to be
|
||||
; used from the Asterisk CLI.
|
||||
34
asterisk/dbsep.conf
Executable file
34
asterisk/dbsep.conf
Executable file
@ -0,0 +1,34 @@
|
||||
#
|
||||
# Configuration file for dbsep.cgi
|
||||
#
|
||||
# The purpose of this file is to provide realtime access to a database,
|
||||
# possibly through ODBC, without needing to load the ODBC drivers into
|
||||
# Asterisk, since there are several backend drivers which are rather
|
||||
# buggy.
|
||||
#
|
||||
# We accomplish this separation by using the res_config_curl realtime
|
||||
# driver to connect to a server running dbsep.cgi (or another, which
|
||||
# implements the same protocol).
|
||||
#
|
||||
# This file contains the information necessary to configure dbsep.cgi.
|
||||
#
|
||||
#
|
||||
# Once installed to a web server, you'll need to preload func_curl.so
|
||||
# and res_config_curl.so in modules.conf and configure extconfig.conf:
|
||||
#
|
||||
# voicemail => curl,http://server/path/to/dbsep.cgi/voicemail
|
||||
# sippeers => curl,http://server/path/to/dbsep.cgi/sippeers
|
||||
#
|
||||
|
||||
# The Data Source Name, as specified by the Perl DBI module.
|
||||
# Typically, this will be along the lines of 'DBI:mysql:astdbname[:dbhostname]' or 'DBI:Pg:dbname=astdbname;hostname=dbhostname'
|
||||
dsn=somedsn
|
||||
|
||||
# Connected database user
|
||||
dbuser=someuser
|
||||
|
||||
# And its password
|
||||
dbpass=password
|
||||
|
||||
# For most databases, this is fine. Set to 'no' for Sybase or MS SQL Server.
|
||||
backslash_is_escape=yes
|
||||
5
asterisk/dnsmgr.conf
Executable file
5
asterisk/dnsmgr.conf
Executable file
@ -0,0 +1,5 @@
|
||||
[general]
|
||||
;enable=yes ; enable creation of managed DNS lookups
|
||||
; default is 'no'
|
||||
;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds
|
||||
; default is 300 (5 minutes)
|
||||
42
asterisk/dsp.conf
Executable file
42
asterisk/dsp.conf
Executable file
@ -0,0 +1,42 @@
|
||||
[default]
|
||||
;
|
||||
; Length of sound (in milliseconds) before a period of silence is considered
|
||||
; to be a change from talking to silence or a period of noise converts silence
|
||||
; to talking. [default=256]
|
||||
;
|
||||
;silencethreshold=256
|
||||
|
||||
; DTMF Reverse Twist and Normal Twist is the difference in power between the row and column energies.
|
||||
;
|
||||
; Normal Twist is where the row energy is greater than the column energy.
|
||||
; Reverse Twist is where the column energy is greater.
|
||||
;
|
||||
; Power level difference between frequencies for different Administrations/RPOAs
|
||||
; Power Gain equiv
|
||||
; normal reverse dB's
|
||||
; AT&T(default) 6.31 2.51 8dB(normal), 4dB(reverse)
|
||||
; NTT 3.16 3.16 Max. 5dB
|
||||
; Danish 3.98 3.98 Max. 6dB
|
||||
; Australian 10.0 10.0 Max. 10dB
|
||||
; Brazilian 7.94 7.94 Max. 9dB
|
||||
; ETSI 3.98 3.98 Max. 6dB
|
||||
|
||||
;previous version compatible AT&T values
|
||||
; RADIO_RELAX disabled, and relaxdtmf=no
|
||||
; 6.30 2.50 7.99dB(normal), 3.98dB(reverse)
|
||||
; RADIO_RELAX disabled, and relaxdtmf=yes
|
||||
; 6.30 4.00 7.99dB(normal), 6.02dB(reverse)
|
||||
; RADIO_RELAX enabled, and relaxdtmf=no
|
||||
; 6.30 2.50 7.99dB(normal), 3.984dB(reverse)
|
||||
; RADIO_RELAX enabled, and relaxdtmf=yes
|
||||
; 6.30 6.50 7.99dB(normal), 8.13dB(reverse)
|
||||
|
||||
;If you don't know what these mean, don't change them.
|
||||
;dtmf_normal_twist=6.31
|
||||
;dtmf_reverse_twist=2.51
|
||||
;relax_dtmf_normal_twist=6.31
|
||||
;relax_dtmf_reverse_twist=3.98
|
||||
|
||||
;successive number hits/misses of 12.75ms before a digit/nodigit is considered valid
|
||||
;dtmf_hits_to_begin=2
|
||||
;dtmf_misses_to_end=3
|
||||
277
asterisk/dundi.conf
Executable file
277
asterisk/dundi.conf
Executable file
@ -0,0 +1,277 @@
|
||||
;
|
||||
; DUNDi configuration file
|
||||
;
|
||||
; For more information about DUNDi, see http://www.dundi.com
|
||||
;
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; The "general" section contains general parameters relating
|
||||
; to the operation of the dundi client and server.
|
||||
;
|
||||
; The first part should be your complete contact information
|
||||
; should someone else in your peer group need to contact you.
|
||||
;
|
||||
;department=Your Department
|
||||
;organization=Your Company, Inc.
|
||||
;locality=Your City
|
||||
;stateprov=ST
|
||||
;country=US
|
||||
;email=your@email.com
|
||||
;phone=+12565551212
|
||||
;
|
||||
;
|
||||
; Specify bind address. IPv6 addresses are accepted. Default is 0.0.0.0
|
||||
; You can specify 'bindaddr2' to bind to another address however
|
||||
; 'bindaddr and 'bindaddr2' need to be different IP protocols.
|
||||
; Specify port number. Default is 4520.
|
||||
;
|
||||
;bindaddr=0.0.0.0
|
||||
;port=4520
|
||||
;
|
||||
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of the tos parameter.
|
||||
;tos=ef
|
||||
;
|
||||
; Our entity identifier. (It should generally be the MAC address of the
|
||||
; machine it's running on. Defaults to the first eth address, but you
|
||||
; can override it here, as long as you set it to the MAC of *something*
|
||||
; you own!) The EID can be overridden by a setting in asterisk.conf
|
||||
; or by setting this option.
|
||||
;
|
||||
;entityid=00:07:E9:3B:76:60
|
||||
;
|
||||
; Peers shall cache our query responses for the specified time
|
||||
; in seconds. Default is 3600.
|
||||
;
|
||||
;cachetime=3600
|
||||
;
|
||||
; This defines the max depth (hops) in which to search the DUNDi system.
|
||||
; Note that the maximum time that we will wait for a response is
|
||||
; (2000 + 200 * ttl) ms.
|
||||
;
|
||||
ttl=32
|
||||
;
|
||||
; If we don't get ACK to our DPDISCOVER within 2000ms and autokill is set
|
||||
; to yes then we cancel the whole thing (that's enough time for one
|
||||
; retransmission only). This is used to keep things from stalling for a long
|
||||
; time for a host that is not available, but would be ill advised for bad
|
||||
; connections. In addition to 'yes' or 'no' you can also specify a number
|
||||
; of milliseconds. See 'qualify' for individual peers to turn on for just
|
||||
; a specific peer.
|
||||
;
|
||||
autokill=yes
|
||||
;
|
||||
; pbx_dundi creates a rotating key called "secret", under the family
|
||||
; 'secretpath'. The default family is dundi (resulting in
|
||||
; the key being held at dundi/secret).
|
||||
;
|
||||
;secretpath=dundi
|
||||
;
|
||||
; The 'storehistory' option (also changeable at runtime with
|
||||
; 'dundi store history on' and 'dundi store history off') will
|
||||
; cause the DUNDi engine to keep track of the last several
|
||||
; queries and the amount of time each query took to execute
|
||||
; for the purpose of tracking slow nodes. This option is
|
||||
; off by default due to performance impacts.
|
||||
;
|
||||
;storehistory=yes
|
||||
|
||||
[mappings]
|
||||
;
|
||||
; The "mappings" section maps DUNDi contexts
|
||||
; to contexts on the local asterisk system. Remember
|
||||
; that numbers that are made available under the e164
|
||||
; DUNDi context are regulated by the DUNDi General Peering
|
||||
; Agreement (GPA) if you are a member of the DUNDi E.164
|
||||
; Peering System.
|
||||
;
|
||||
; dundi_context => [local_context,weight,tech,dest{,options}]
|
||||
;
|
||||
; 'dundi_context' is the name of the context being requested
|
||||
; within the DUNDi request
|
||||
;
|
||||
; 'local_context' is the name of the context on the local system
|
||||
; in which numbers can be looked up for which responses shall be given.
|
||||
;
|
||||
; 'weight' is the weight to use for the responses provided from this
|
||||
; mapping. The number must be >= 0 and < 60000. Since it is totally
|
||||
; valid to receive multiple responses to a query, responses received
|
||||
; with a lower weight are tried first. Note that the weight has a
|
||||
; special meaning in the e164 context - see the GPA for more details.
|
||||
;
|
||||
; 'tech' is the technology to use (IAX2, SIP, H323)
|
||||
;
|
||||
; 'dest' is the Dial application's channel technology resource destination
|
||||
; to supply for reaching that number. The following variables can be used
|
||||
; in the destination string and will be automatically substituted:
|
||||
; ${NUMBER}: The number being requested
|
||||
; ${IPADDR}: The IP address to connect to
|
||||
; ${SECRET}: The current IAX2 rotating secret key to be used
|
||||
;
|
||||
; Further options may include:
|
||||
;
|
||||
; nounsolicited: No unsolicited calls of any type permitted via this
|
||||
; route
|
||||
; nocomunsolicit: No commercial unsolicited calls permitted via
|
||||
; this route
|
||||
; residential: This number is known to be a residence
|
||||
; commercial: This number is known to be a business
|
||||
; mobile: This number is known to be a mobile phone
|
||||
; nocomunsolicit: No commercial unsolicited calls permitted via
|
||||
; this route
|
||||
; nopartial: Do not search for partial matches
|
||||
;
|
||||
; There *must* exist an entry in mappings for DUNDi to respond
|
||||
; to any request, although it may be empty.
|
||||
;
|
||||
;empty_context =>
|
||||
;
|
||||
;e164 => dundi-e164-canonical,0,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
|
||||
;e164 => dundi-e164-customers,100,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
|
||||
;e164 => dundi-e164-via-pstn,400,IAX2,dundi:${SECRET}@${IPADDR}/${NUMBER},nounsolicited,nocomunsolicit,nopartial
|
||||
;
|
||||
;digexten => default,0,IAX2,guest@lappy/${NUMBER}
|
||||
|
||||
;
|
||||
; Weights for mappings can be set a few different ways:
|
||||
;
|
||||
; 1) It can be set as a static number.
|
||||
;testmap1 => context1,222,IAX2,guest@peer1/${NUMBER}
|
||||
;
|
||||
; 2) It can be an Asterisk global variable.
|
||||
;testmap2 => context2,${DUNDITESTVAR},IAX2,guest@peer2${NUMBER}
|
||||
;
|
||||
; 3) It can be retrieved using a dialplan function. This can be extremely
|
||||
; useful if you want to let an external script decide what the weight
|
||||
; in a response should be.
|
||||
;testmap3 => context3,${SHELL(echo 123)},IAX2,guest@peer3/${NUMBER}
|
||||
;
|
||||
; The built in variables ${SECRET}, ${IPADDR} and ${NUMBER} can also be
|
||||
; passed to the weight. For example, you could pass the ${NUMBER} value
|
||||
; to your SHELL() script and use that to dynamically return a weight.
|
||||
;
|
||||
; Note when using a global variable or dialplan function to set the
|
||||
; weight for a mapping that response caching should be disabled if you
|
||||
; plan for these values to change frequently at all. If the results are
|
||||
; cached then any change in value will not take effect until the cache
|
||||
; has expired.
|
||||
;
|
||||
|
||||
;
|
||||
; The remaining sections represent the peers that we fundamentally trust.
|
||||
; The section name specifies the peer's entityid. You can specify which
|
||||
; DUNDi contexts with which you want the trust to be established.
|
||||
;
|
||||
; inkey - What key they will be authenticating to us with
|
||||
;
|
||||
; outkey - What key we use to authenticate to them
|
||||
;
|
||||
; host - What their host is (DNS name, IP address, or dynamic)
|
||||
;
|
||||
; port - The port where their host is listening (default: 4520)
|
||||
;
|
||||
; ustothem - Explicitly specify the entityid we use with this peer.
|
||||
;
|
||||
; order - What search order to use. May be 'primary', 'secondary',
|
||||
; 'tertiary' or 'quartiary'. In large systems, it is beneficial
|
||||
; to only query one up-stream host in order to maximize caching
|
||||
; value. Adding one with primary and one with secondary gives you
|
||||
; redundancy without sacrificing performance.
|
||||
;
|
||||
; include - Includes this peer when searching a particular context
|
||||
; for lookup (set "all" to perform all lookups with that
|
||||
; host. This is also the context in which peers are permitted
|
||||
; to precache.
|
||||
;
|
||||
; noinclude - Disincludes this peer when searching a particular context
|
||||
; for lookup (set "all" to perform no lookups with that
|
||||
; host.
|
||||
;
|
||||
; permit - Permits this peer to search a given DUNDi context on
|
||||
; the local system. Set "all" to permit this host to
|
||||
; lookup all contexts. This is also a context for which
|
||||
; we will create/forward PRECACHE commands.
|
||||
;
|
||||
; deny - Denies this peer to search a given DUNDi context on
|
||||
; the local system. Set "all" to deny this host to
|
||||
; lookup all contexts.
|
||||
;
|
||||
; model - inbound, outbound, or symmetric for whether we receive
|
||||
; requests only, transmit requests only, or do both.
|
||||
;
|
||||
; precache - Utilize/Permit precaching with this peer (to pre
|
||||
; cache means to provide an answer when no request
|
||||
; was made and is used so that machines with few
|
||||
; routes can push those routes up to a higher level).
|
||||
; outgoing means we send precache routes to this peer,
|
||||
; incoming means we permit this peer to send us
|
||||
; precache routes. symmetric means we do both.
|
||||
;
|
||||
; Note: You cannot mix symmetric/outbound model with symmetric/inbound
|
||||
; precache, nor can you mix symmetric/inbound model with symmetric/outbound
|
||||
; precache.
|
||||
;
|
||||
; qualify - Enable qualifying the peer to determine reachable status.
|
||||
; Set to yes, no, or number of milliseconds for qualifying
|
||||
; the peer's reachable status.
|
||||
;
|
||||
; register - Enable registering with the peer. This presupposes that the
|
||||
; peer's host option for us is dynamic. (yes/no value)
|
||||
;
|
||||
; The '*' peer is special and matches an unspecified entity
|
||||
;
|
||||
|
||||
;
|
||||
; Sample Primary e164 DUNDi peer
|
||||
;
|
||||
;[00:50:8B:F3:75:BB]
|
||||
;model = symmetric
|
||||
;host = 64.215.96.114
|
||||
;inkey = digium
|
||||
;outkey = misery
|
||||
;include = e164
|
||||
;permit = e164
|
||||
;qualify = yes
|
||||
|
||||
;
|
||||
; Sample Secondary e164 DUNDi peer
|
||||
;
|
||||
;[00:A0:C9:96:92:84]
|
||||
;model = symmetric
|
||||
;host = misery.digium.com
|
||||
;inkey = misery
|
||||
;outkey = ourkey
|
||||
;include = e164
|
||||
;permit = e164
|
||||
;qualify = yes
|
||||
;order = secondary
|
||||
|
||||
;
|
||||
; Sample "push mode" downstream host
|
||||
;
|
||||
;[00:0C:76:96:75:28]
|
||||
;model = inbound
|
||||
;host = dynamic
|
||||
;precache = inbound
|
||||
;inkey = littleguy
|
||||
;outkey = ourkey
|
||||
;include = e164 ; In this case used only for precaching
|
||||
;permit = e164
|
||||
;qualify = yes
|
||||
|
||||
;
|
||||
; Sample "push mode" upstream host
|
||||
;
|
||||
;[00:07:E9:3B:76:60]
|
||||
;model = outbound
|
||||
;precache = outbound
|
||||
;host = 216.207.245.34
|
||||
;register = yes
|
||||
;inkey = dhcp34
|
||||
;permit = all ; In this case used only for precaching
|
||||
;include = all
|
||||
;qualify = yes
|
||||
;outkey=foo
|
||||
|
||||
;[*]
|
||||
;
|
||||
22
asterisk/enum.conf
Executable file
22
asterisk/enum.conf
Executable file
@ -0,0 +1,22 @@
|
||||
;
|
||||
; ENUM Configuration for resolving phone numbers over DNS
|
||||
;
|
||||
; Sample config for Asterisk
|
||||
; This file is reloaded at "module reload enum" in the CLI
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; The search list for domains may be customized. Domains are searched
|
||||
; in the order they are listed here.
|
||||
;
|
||||
search => e164.arpa
|
||||
;
|
||||
; If you'd like to use the E.164.org public ENUM registry in addition
|
||||
; to the official e164.arpa one, uncomment the following line
|
||||
;
|
||||
;search => e164.org
|
||||
;
|
||||
; As there are more H323 drivers available you have to select to which
|
||||
; drive a H323 URI will map. Default is "H323".
|
||||
;
|
||||
h323driver => H323
|
||||
112
asterisk/extconfig.conf
Executable file
112
asterisk/extconfig.conf
Executable file
@ -0,0 +1,112 @@
|
||||
;
|
||||
; Static and realtime external configuration
|
||||
; engine configuration
|
||||
;
|
||||
; See https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
|
||||
; for basic table formatting information.
|
||||
;
|
||||
[settings]
|
||||
;
|
||||
; Static configuration files:
|
||||
;
|
||||
; file.conf => driver,database[,table[,priority]]
|
||||
;
|
||||
; maps a particular configuration file to the given
|
||||
; database driver, database and table (or uses the
|
||||
; name of the file as the table if not specified)
|
||||
;
|
||||
; Uncomment to load queues.conf via the odbc engine.
|
||||
;
|
||||
;queues.conf => odbc,asterisk,ast_config
|
||||
;extensions.conf => sqlite,asterisk,ast_config
|
||||
;
|
||||
; The following files CANNOT be loaded from Realtime storage:
|
||||
; asterisk.conf
|
||||
; extconfig.conf (this file)
|
||||
; logger.conf
|
||||
;
|
||||
; Additionally, the following files cannot be loaded from
|
||||
; Realtime storage unless the storage driver is loaded
|
||||
; early using 'preload' statements in modules.conf:
|
||||
; manager.conf
|
||||
; cdr.conf
|
||||
; rtp.conf
|
||||
;
|
||||
; Named ACLs specified in realtime also can not be used
|
||||
; from manager.conf unless the storage driver is preloaded.
|
||||
; Attempting to use a realtime stored named ACL before the
|
||||
; driver is loaded will result in an invalid ACL which
|
||||
; rejects all addresses.
|
||||
;
|
||||
; Realtime configuration engine
|
||||
;
|
||||
; maps a particular family of realtime
|
||||
; configuration to a given database driver,
|
||||
; database and table (or uses the name of
|
||||
; the family if the table is not specified
|
||||
;
|
||||
;example => odbc,asterisk,alttable,1
|
||||
;example => mysql,asterisk,alttable,2
|
||||
;example2 => ldap,"dc=oxymium,dc=net",example2
|
||||
;
|
||||
; Additionally, priorities are now supported for use as failover methods
|
||||
; for retrieving realtime data. If one connection fails to retrieve any
|
||||
; information, the next sequential priority will be tried next. This
|
||||
; especially works well with ODBC connections, since res_odbc now caches
|
||||
; when connection failures occur and prevents immediately retrying those
|
||||
; connections until after a specified timeout. Note: priorities must
|
||||
; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
|
||||
; and 4, then 4 will be ignored, because there is no 3).
|
||||
;
|
||||
;
|
||||
; Possible driver backends:
|
||||
;
|
||||
; "odbc" is shown in the examples below, but is not the only valid realtime
|
||||
; engine. Here are several of the possible options:
|
||||
; odbc ... res_config_odbc
|
||||
; sqlite ... res_config_sqlite
|
||||
; sqlite3 ... res_config_sqlite3
|
||||
; pgsql ... res_config_pgsql
|
||||
; curl ... res_config_curl
|
||||
; ldap ... res_config_ldap
|
||||
; mysql ... res_config_mysql (available via add-ons in menuselect)
|
||||
;
|
||||
; Note: The res_config_pgsql and res_config_sqlite backends configure the
|
||||
; database used in their respective configuration files and ignore the
|
||||
; database name configured in this file.
|
||||
;
|
||||
;iaxusers => odbc,asterisk
|
||||
;iaxpeers => odbc,asterisk
|
||||
;sippeers => odbc,asterisk
|
||||
;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)
|
||||
;ps_endpoints => odbc,asterisk
|
||||
;ps_auths => odbc,asterisk
|
||||
;ps_aors => odbc,asterisk
|
||||
;ps_domain_aliases => odbc,asterisk
|
||||
;ps_endpoint_id_ips => odbc,asterisk
|
||||
;ps_outbound_publishes => odbc,asterisk
|
||||
;ps_inbound_publications = odbc,asterisk
|
||||
;ps_asterisk_publications = odbc,asterisk
|
||||
;voicemail => odbc,asterisk
|
||||
;extensions => odbc,asterisk
|
||||
;meetme => mysql,general
|
||||
;queues => odbc,asterisk
|
||||
;queue_members => odbc,asterisk
|
||||
;queue_rules => odbc,asterisk
|
||||
;acls => odbc,asterisk
|
||||
;musiconhold => mysql,general
|
||||
;musiconhold_entry => mysql,general
|
||||
;queue_log => mysql,general
|
||||
;
|
||||
;
|
||||
; While most dynamic realtime engines are automatically used when defined in
|
||||
; this file, 'extensions', distinctively, is not. To activate dynamic realtime
|
||||
; extensions, you must turn them on in each respective context within
|
||||
; extensions.conf with a switch statement. The syntax is:
|
||||
; switch => Realtime/[[db_context@]tablename]/<opts>
|
||||
; The only option available currently is the 'p' option, which disallows
|
||||
; extension pattern queries to the database. If you have no patterns defined
|
||||
; in a particular context, this will save quite a bit of CPU time. However,
|
||||
; note that using dynamic realtime extensions is not recommended anymore as a
|
||||
; best practice; instead, you should consider writing a static dialplan with
|
||||
; proper data abstraction via a tool like func_odbc.
|
||||
456
asterisk/extensions.ael
Executable file
456
asterisk/extensions.ael
Executable file
@ -0,0 +1,456 @@
|
||||
//
|
||||
// Example AEL config file
|
||||
//
|
||||
//
|
||||
// Static extension configuration file, used by
|
||||
// the pbx_ael module. This is where you configure all your
|
||||
// inbound and outbound calls in Asterisk.
|
||||
//
|
||||
// This configuration file is reloaded
|
||||
// - With the "ael reload" command in the CLI
|
||||
// - With the "reload" command (that reloads everything) in the CLI
|
||||
|
||||
// The "Globals" category contains global variables that can be referenced
|
||||
// in the dialplan by using the GLOBAL dialplan function:
|
||||
// ${GLOBAL(VARIABLE)}
|
||||
// ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
|
||||
// Unix/Linux environmental variables are reached with the ENV dialplan
|
||||
// function: ${ENV(VARIABLE)}
|
||||
//
|
||||
|
||||
// NOTE! NOTE! NOTE!
|
||||
// Asterisk by default will load both extensions.conf and extensions.ael files.
|
||||
// Upon loading these files the dialplans generated from both with be merged,
|
||||
// so you must make sure that you don't have any overlapping contexts or global
|
||||
// variables. If you do, then unexpected behavior may result when the data is
|
||||
// merged.
|
||||
// NOTE! NOTE! NOTE!
|
||||
|
||||
globals {
|
||||
CONSOLE-AEL="Console/dsp"; // Console interface for demo
|
||||
//CONSOLE-AEL=Zap/1;
|
||||
//CONSOLE-AEL=Phone/phone0;
|
||||
IAXINFO-AEL=guest; // IAXtel username/password
|
||||
//IAXINFO-AEL="myuser:mypass";
|
||||
OUTBOUND-TRUNK="Zap/g2"; // Trunk interface
|
||||
//
|
||||
// Note the 'g2' in the OUTBOUND-TRUNK variable above. It specifies which group (defined
|
||||
// in chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use in
|
||||
// the specified group. The four possible options are:
|
||||
//
|
||||
// g: select the lowest-numbered non-busy DAHDI channel
|
||||
// (aka. ascending sequential hunt group).
|
||||
// G: select the highest-numbered non-busy DAHDI channel
|
||||
// (aka. descending sequential hunt group).
|
||||
// r: use a round-robin search, starting at the next highest channel than last
|
||||
// time (aka. ascending rotary hunt group).
|
||||
// R: use a round-robin search, starting at the next lowest channel than last
|
||||
// time (aka. descending rotary hunt group).
|
||||
//
|
||||
OUTBOUND-TRUNKMSD=1; // MSD digits to strip (usually 1 or 0)
|
||||
//OUTBOUND-TRUNK2=IAX2/user:pass@provider;
|
||||
};
|
||||
|
||||
//
|
||||
// Any category other than "General" and "Globals" represent
|
||||
// extension contexts, which are collections of extensions.
|
||||
//
|
||||
// Extension names may be numbers, letters, or combinations
|
||||
// thereof. If an extension name is prefixed by a '_'
|
||||
// character, it is interpreted as a pattern rather than a
|
||||
// literal. In patterns, some characters have special meanings:
|
||||
//
|
||||
// X - any digit from 0-9
|
||||
// Z - any digit from 1-9
|
||||
// N - any digit from 2-9
|
||||
// [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
|
||||
// . - wildcard, matches anything remaining (e.g. _9011. matches
|
||||
// anything starting with 9011 excluding 9011 itself)
|
||||
// ! - wildcard, causes the matching process to complete as soon as
|
||||
// it can unambiguously determine that no other matches are possible
|
||||
//
|
||||
// For example the extension _NXXXXXX would match normal 7 digit dialings,
|
||||
// while _1NXXNXXXXXX would represent an area code plus phone number
|
||||
// preceded by a one.
|
||||
//
|
||||
// Each step of an extension is ordered by priority, which must
|
||||
// always start with 1 to be considered a valid extension. The priority
|
||||
// "next" or "n" means the previous priority plus one, regardless of whether
|
||||
// the previous priority was associated with the current extension or not.
|
||||
// The priority "same" or "s" means the same as the previously specified
|
||||
// priority, again regardless of whether the previous entry was for the
|
||||
// same extension. Priorities may be immediately followed by a plus sign
|
||||
// and another integer to add that amount (most useful with 's' or 'n').
|
||||
// Priorities may then also have an alias, or label, in
|
||||
// parenthesis after their name which can be used in goto situations
|
||||
//
|
||||
// Contexts contain several lines, one for each step of each
|
||||
// extension, which can take one of two forms as listed below,
|
||||
// with the first form being preferred. One may include another
|
||||
// context in the current one as well, optionally with a
|
||||
// date and time. Included contexts are included in the order
|
||||
// they are listed.
|
||||
//
|
||||
//context name {
|
||||
// exten-name => {
|
||||
// application(arg1,arg2,...);
|
||||
//
|
||||
// Timing list for includes is
|
||||
//
|
||||
// <time range>|<days of week>|<days of month>|<months>
|
||||
//
|
||||
// includes {
|
||||
// daytime|9:00-17:00|mon-fri|*|*;
|
||||
// };
|
||||
//
|
||||
// ignorepat can be used to instruct drivers to not cancel dialtone upon
|
||||
// receipt of a particular pattern. The most commonly used example is
|
||||
// of course '9' like this:
|
||||
//
|
||||
// ignorepat => 9;
|
||||
//
|
||||
// so that dialtone remains even after dialing a 9.
|
||||
//};
|
||||
|
||||
|
||||
//
|
||||
// Sample entries for extensions.conf
|
||||
//
|
||||
//
|
||||
context ael-dundi-e164-canonical {
|
||||
//
|
||||
// List canonical entries here
|
||||
//
|
||||
// 12564286000 => &ael-std-exten(6000,IAX2/foo);
|
||||
// _125642860XX => Dial(IAX2/otherbox/${EXTEN:7});
|
||||
};
|
||||
|
||||
context ael-dundi-e164-customers {
|
||||
//
|
||||
// If you are an ITSP or Reseller, list your customers here.
|
||||
//
|
||||
//_12564286000 => Dial(SIP/customer1);
|
||||
//_12564286001 => Dial(IAX2/customer2);
|
||||
};
|
||||
|
||||
context ael-dundi-e164-via-pstn {
|
||||
//
|
||||
// If you are freely delivering calls to the PSTN, list them here
|
||||
//
|
||||
//_1256428XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Expose all of 256-428
|
||||
//_1256325XXXX => Dial(DAHDI/G2/${EXTEN:7}); // Ditto for 256-325
|
||||
};
|
||||
|
||||
context ael-dundi-e164-local {
|
||||
//
|
||||
// Context to put your dundi IAX2 or SIP user in for
|
||||
// full access
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-canonical;
|
||||
ael-dundi-e164-customers;
|
||||
ael-dundi-e164-via-pstn;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-dundi-e164-switch {
|
||||
//
|
||||
// Just a wrapper for the switch
|
||||
//
|
||||
|
||||
switches {
|
||||
DUNDi/e164;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-dundi-e164-lookup {
|
||||
//
|
||||
// Locally to lookup, try looking for a local E.164 solution
|
||||
// then try DUNDi if we don't have one.
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-local;
|
||||
ael-dundi-e164-switch;
|
||||
};
|
||||
//
|
||||
};
|
||||
|
||||
//
|
||||
// DUNDi can also be implemented as a Macro instead of using
|
||||
// the Local channel driver.
|
||||
//
|
||||
macro ael-dundi-e164(exten) {
|
||||
//
|
||||
// ARG1 is the extension to Dial
|
||||
//
|
||||
goto ${exten}|1;
|
||||
return;
|
||||
};
|
||||
|
||||
//
|
||||
// Here are the entries you need to participate in the IAXTEL
|
||||
// call routing system. Most IAXTEL numbers begin with 1-700, but
|
||||
// there are exceptions. For more information, and to sign
|
||||
// up, please go to www.gnophone.com or www.iaxtel.com
|
||||
//
|
||||
context ael-iaxtel700 {
|
||||
_91700XXXXXXX => Dial(IAX2/${IAXINFO-AEL}@iaxtel.com/${EXTEN:1}@iaxtel);
|
||||
};
|
||||
|
||||
//
|
||||
// The SWITCH statement permits a server to share the dialplan with
|
||||
// another server. Use with care: Reciprocal switch statements are not
|
||||
// allowed (e.g. both A -> B and B -> A), and the switched server needs
|
||||
// to be on-line or else dialing can be severly delayed.
|
||||
//
|
||||
context ael-iaxprovider {
|
||||
switches {
|
||||
// IAX2/user:[key]@myserver/mycontext;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunkint {
|
||||
//
|
||||
// International long distance through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_9011. => {
|
||||
&ael-dundi-e164(${EXTEN:4});
|
||||
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunkld {
|
||||
//
|
||||
// Long distance context accessed through trunk
|
||||
//
|
||||
includes {
|
||||
ael-dundi-e164-lookup;
|
||||
};
|
||||
_91NXXNXXXXXX => {
|
||||
&ael-dundi-e164(${EXTEN:1});
|
||||
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunklocal {
|
||||
//
|
||||
// Local seven-digit dialing accessed through trunk interface
|
||||
//
|
||||
_9NXXXXXX => {
|
||||
Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
};
|
||||
};
|
||||
|
||||
context ael-trunktollfree {
|
||||
//
|
||||
// Long distance context accessed through trunk interface
|
||||
//
|
||||
|
||||
_91800NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
_91888NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
_91877NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
_91866NXXXXXX => Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}});
|
||||
};
|
||||
|
||||
context ael-international {
|
||||
//
|
||||
// Master context for international long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-longdistance;
|
||||
ael-trunkint;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-longdistance {
|
||||
//
|
||||
// Master context for long distance
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-local;
|
||||
ael-trunkld;
|
||||
};
|
||||
};
|
||||
|
||||
context ael-local {
|
||||
//
|
||||
// Master context for local, toll-free, and iaxtel calls only
|
||||
//
|
||||
ignorepat => 9;
|
||||
includes {
|
||||
ael-default;
|
||||
ael-trunklocal;
|
||||
ael-iaxtel700;
|
||||
ael-trunktollfree;
|
||||
ael-iaxprovider;
|
||||
};
|
||||
};
|
||||
|
||||
//
|
||||
// You can use an alternative switch type as well, to resolve
|
||||
// extensions that are not known here, for example with remote
|
||||
// IAX switching you transparently get access to the remote
|
||||
// Asterisk PBX
|
||||
//
|
||||
// switch => IAX2/user:password@bigserver/local
|
||||
//
|
||||
// An "lswitch" is like a switch but is literal, in that
|
||||
// variable substitution is not performed at load time
|
||||
// but is passed to the switch directly (presumably to
|
||||
// be substituted in the switch routine itself)
|
||||
//
|
||||
// lswitch => Loopback/12${EXTEN}@othercontext
|
||||
//
|
||||
// An "eswitch" is like a switch but the evaluation of
|
||||
// variable substitution is performed at runtime before
|
||||
// being passed to the switch routine.
|
||||
//
|
||||
// eswitch => IAX2/context@${CURSERVER}
|
||||
|
||||
|
||||
macro ael-std-exten-ael( ext , dev ) {
|
||||
Dial(${dev}/${ext},20);
|
||||
switch(${DIALSTATUS}) {
|
||||
case BUSY:
|
||||
Voicemail(${ext},b);
|
||||
break;
|
||||
default:
|
||||
Voicemail(${ext},u);
|
||||
};
|
||||
catch a {
|
||||
VoiceMailMain(${ext});
|
||||
return;
|
||||
};
|
||||
return;
|
||||
};
|
||||
|
||||
context ael-demo {
|
||||
s => {
|
||||
Wait(1);
|
||||
Answer();
|
||||
Set(TIMEOUT(digit)=5);
|
||||
Set(TIMEOUT(response)=10);
|
||||
restart:
|
||||
Background(demo-congrats);
|
||||
instructions:
|
||||
for (x=0; ${x} < 3; x=${x} + 1) {
|
||||
Background(demo-instruct);
|
||||
WaitExten();
|
||||
};
|
||||
};
|
||||
2 => {
|
||||
Background(demo-moreinfo);
|
||||
goto s|instructions;
|
||||
};
|
||||
3 => {
|
||||
Set(LANGUAGE()=fr);
|
||||
goto s|restart;
|
||||
};
|
||||
1000 => {
|
||||
goto ael-default|s|1;
|
||||
};
|
||||
500 => {
|
||||
Playback(demo-abouttotry);
|
||||
Dial(IAX2/guest@misery.digium.com/s@default);
|
||||
Playback(demo-nogo);
|
||||
goto s|instructions;
|
||||
};
|
||||
600 => {
|
||||
Playback(demo-echotest);
|
||||
Echo();
|
||||
Playback(demo-echodone);
|
||||
goto s|instructions;
|
||||
};
|
||||
_1234 => &ael-std-exten-ael(${EXTEN}, "IAX2");
|
||||
8500 => {
|
||||
VoicemailMain();
|
||||
goto s|instructions;
|
||||
};
|
||||
# => {
|
||||
Playback(demo-thanks);
|
||||
Hangup();
|
||||
};
|
||||
t => goto #|1;
|
||||
i => Playback(invalid);
|
||||
};
|
||||
|
||||
|
||||
//
|
||||
// If you wish to use AEL for your default context, remove it
|
||||
// from extensions.conf (or change its name or comment it out)
|
||||
// and then uncomment the one here.
|
||||
//
|
||||
|
||||
context ael-default {
|
||||
|
||||
// By default we include the demo. In a production system, you
|
||||
// probably don't want to have the demo there.
|
||||
|
||||
includes {
|
||||
ael-demo;
|
||||
};
|
||||
//
|
||||
// Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
|
||||
// Note that you must have a [sipprovider] section in sip.conf whereas
|
||||
// the otherprovider.net example does not require such a peer definition
|
||||
//
|
||||
//_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);
|
||||
//_42X. => Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT);
|
||||
|
||||
// Real extensions would go here. Generally you want real extensions to be
|
||||
// 4 or 5 digits long (although there is no such requirement) and start with a
|
||||
// single digit that is fairly large (like 6 or 7) so that you have plenty of
|
||||
// room to overlap extensions and menu options without conflict. You can alias
|
||||
// them with names, too, and use global variables
|
||||
|
||||
// 6245 => {
|
||||
// hint(SIP/Grandstream1&SIP/Xlite1,Joe Schmoe); // Channel hints for presence
|
||||
// Dial(SIP/Grandstream1,20,rt); // permit transfer
|
||||
// Dial(${HINT}/5245},20,rtT); // Use hint as listed
|
||||
// switch(${DIALSTATUS}) {
|
||||
// case BUSY:
|
||||
// Voicemail(6245,b);
|
||||
// return;
|
||||
// default:
|
||||
// Voicemail(6245,u);
|
||||
// return;
|
||||
// };
|
||||
// };
|
||||
|
||||
// 6361 => Dial(IAX2/JaneDoe,,rm); // ring without time limit
|
||||
// 6389 => Dial(MGCP/aaln/1@192.168.0.14);
|
||||
// 6394 => Dial(Local/6275/n); // this will dial ${MARK}
|
||||
|
||||
// 6275 => &ael-stdexten(6275,${MARK}); // assuming ${MARK} is something like DAHDI/2
|
||||
// mark => goto 6275|1; // alias mark to 6275
|
||||
// 6536 => &ael-stdexten(6236,${WIL}); // Ditto for wil
|
||||
// wil => goto 6236|1;
|
||||
//
|
||||
// Some other handy things are an extension for checking voicemail via
|
||||
// voicemailmain
|
||||
//
|
||||
// 8500 => {
|
||||
// VoicemailMain();
|
||||
// Hangup();
|
||||
// };
|
||||
//
|
||||
// Or a conference room (you'll need to edit meetme.conf to enable this room)
|
||||
//
|
||||
// 8600 => Meetme(1234);
|
||||
//
|
||||
// Or playing an announcement to the called party, as soon it answers
|
||||
//
|
||||
// 8700 => Dial(${MARK},30,A(/path/to/my/announcemsg))
|
||||
//
|
||||
// For more information on applications, just type "show applications" at your
|
||||
// friendly Asterisk CLI prompt.
|
||||
//
|
||||
// 'show application <command>' will show details of how you
|
||||
// use that particular application in this file, the dial plan.
|
||||
//
|
||||
}
|
||||
52
asterisk/extensions.conf
Executable file
52
asterisk/extensions.conf
Executable file
@ -0,0 +1,52 @@
|
||||
;-------------------------------------------------------
|
||||
; General Section
|
||||
;-------------------------------------------------------
|
||||
[general]
|
||||
static=yes
|
||||
writeprotect=yes
|
||||
clearglobalvars=yes
|
||||
autofallthrough=yes
|
||||
|
||||
[from-internal]
|
||||
exten => 200,1,Stasis(incomingCall)
|
||||
exten => 200,n,Dial(SIP/domofon,120)
|
||||
exten => 200,n,Hangup()
|
||||
|
||||
exten => 203,1,Stasis(incomingCall)
|
||||
exten => 203,n,Dial(SIP/domofon-mohovaya,120)
|
||||
exten => 203,n,Hangup()
|
||||
|
||||
[from-external]
|
||||
exten => 200,1,Stasis(incomingCall)
|
||||
exten => 200,n,Dial(PJSIP/domofon,120)
|
||||
exten => 200,n,Hangup()
|
||||
|
||||
exten => 203,1,Stasis(incomingCall)
|
||||
exten => 203,n,Dial(PJSIP/domofon-mohovaya,120)
|
||||
exten => 203,n,Hangup()
|
||||
|
||||
exten => 201,1,Stasis(incomingCall)
|
||||
exten => 201,n,Dial(PJSIP/domofon-demo,120)
|
||||
exten => 201,n,Hangup()
|
||||
|
||||
exten => 15,1,Dial(PJSIP/15, 120)
|
||||
|
||||
exten => 16,1,Dial(PJSIP/16, 120)
|
||||
|
||||
;[internal]
|
||||
;exten => 15,1,Dial(PJSIP/15, 120)
|
||||
|
||||
;[invalid]
|
||||
[internal]
|
||||
exten => _X,1,Stasis(hello)
|
||||
exten => _X,n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
exten => _X,n,Hangup
|
||||
exten => _XX,1,Stasis(hello)
|
||||
exten => _XX,n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
exten => _XX,n,Hangup
|
||||
exten => _XXX,1,Stasis(hello)
|
||||
exten => _XXX,n,Dial(PJSIP/${EXTEN}&PJSIP/comfortech/${EXTEN},120)
|
||||
exten => _XXX,n,Hangup
|
||||
|
||||
[default]
|
||||
|
||||
74
asterisk/extensions.conf.default
Executable file
74
asterisk/extensions.conf.default
Executable file
@ -0,0 +1,74 @@
|
||||
;-------------------------------------------------------
|
||||
; General Section
|
||||
;-------------------------------------------------------
|
||||
[general]
|
||||
context=default
|
||||
transport=udp
|
||||
allowguest=no
|
||||
allowoverlap=yes
|
||||
udpbindaddr=::
|
||||
tcpenable=no
|
||||
tcpbindaddr=::
|
||||
srvlookup=yes
|
||||
disallow=all
|
||||
allow=alaw
|
||||
callcounter=yes
|
||||
limitonpeer=yes
|
||||
promiscredir=no
|
||||
dtmfmode=rfc2833
|
||||
rtcachefriends=yes
|
||||
callevents=yes
|
||||
language=ru
|
||||
videosupport=yes
|
||||
|
||||
register => domofon-9line6:w1UDBBEvCFtykUjs@sip2.comfortel.pro
|
||||
|
||||
[comfortel]
|
||||
type=friend
|
||||
disallow=all
|
||||
allow=alaw
|
||||
call-limit=1
|
||||
context=outgoing
|
||||
host=sip2.comfortel.pro
|
||||
directmedia=no
|
||||
qualify=yes
|
||||
nat=force_rport,comedia
|
||||
fromdomain=sip.comfortel.pro
|
||||
secret=w1UDBBEvCFtykUjs
|
||||
defaultuser=domofon-9line6
|
||||
fromuser=domofon-9line6
|
||||
|
||||
[user](!)
|
||||
type=friend
|
||||
context=invalid
|
||||
secret=temtikDodd
|
||||
allow=all
|
||||
host=dynamic
|
||||
directmedia=no
|
||||
qualify=yes
|
||||
;nat=force_rport,comedia
|
||||
|
||||
|
||||
[00](user)
|
||||
secret=yavGitujip
|
||||
callerid=100
|
||||
|
||||
[15](user)
|
||||
secret=nixhivsyib
|
||||
callerid=101
|
||||
|
||||
[19](user)
|
||||
secret=roucuffId8
|
||||
callerid=102
|
||||
|
||||
[20](user)
|
||||
secret=yaz8bopGed
|
||||
callerid=103
|
||||
|
||||
[21](user)
|
||||
secret=CrecBebrEm
|
||||
callerid=104
|
||||
|
||||
[domofon](user)
|
||||
secret=VutcaflojUt4driamGic
|
||||
context=local
|
||||
240
asterisk/extensions.lua
Executable file
240
asterisk/extensions.lua
Executable file
@ -0,0 +1,240 @@
|
||||
|
||||
|
||||
CONSOLE = "Console/dsp" -- Console interface for demo
|
||||
--CONSOLE = "DAHDI/1"
|
||||
--CONSOLE = "Phone/phone0"
|
||||
|
||||
IAXINFO = "guest" -- IAXtel username/password
|
||||
--IAXINFO = "myuser:mypass"
|
||||
|
||||
TRUNK = "DAHDI/G2"
|
||||
TRUNKMSD = 1
|
||||
-- TRUNK = "IAX2/user:pass@provider"
|
||||
|
||||
|
||||
--
|
||||
-- Extensions are expected to be defined in a global table named 'extensions'.
|
||||
-- The 'extensions' table should have a group of tables in it, each
|
||||
-- representing a context. Extensions are defined in each context. See below
|
||||
-- for examples.
|
||||
--
|
||||
-- Extension names may be numbers, letters, or combinations thereof. If
|
||||
-- an extension name is prefixed by a '_' character, it is interpreted as
|
||||
-- a pattern rather than a literal. In patterns, some characters have
|
||||
-- special meanings:
|
||||
--
|
||||
-- X - any digit from 0-9
|
||||
-- Z - any digit from 1-9
|
||||
-- N - any digit from 2-9
|
||||
-- [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
|
||||
-- . - wildcard, matches anything remaining (e.g. _9011. matches
|
||||
-- anything starting with 9011 excluding 9011 itself)
|
||||
-- ! - wildcard, causes the matching process to complete as soon as
|
||||
-- it can unambiguously determine that no other matches are possible
|
||||
--
|
||||
-- For example the extension _NXXXXXX would match normal 7 digit
|
||||
-- dialings, while _1NXXNXXXXXX would represent an area code plus phone
|
||||
-- number preceded by a one.
|
||||
--
|
||||
-- If your extension has special characters in it such as '.' and '!' you must
|
||||
-- explicitly make it a string in the tabale definition:
|
||||
--
|
||||
-- ["_special."] = function;
|
||||
-- ["_special!"] = function;
|
||||
--
|
||||
-- There are no priorities. All extensions to asterisk appear to have a single
|
||||
-- priority as if they consist of a single priority.
|
||||
--
|
||||
-- Each context is defined as a table in the extensions table. The
|
||||
-- context names should be strings.
|
||||
--
|
||||
-- One context may be included in another context using the 'includes'
|
||||
-- extension. This extension should be set to a table containing a list
|
||||
-- of context names. Do not put references to tables in the includes
|
||||
-- table.
|
||||
--
|
||||
-- include = {"a", "b", "c"};
|
||||
--
|
||||
-- Channel variables can be accessed thorugh the global 'channel' table.
|
||||
--
|
||||
-- v = channel.var_name
|
||||
-- v = channel["var_name"]
|
||||
-- v.value
|
||||
-- v:get()
|
||||
--
|
||||
-- channel.var_name = "value"
|
||||
-- channel["var_name"] = "value"
|
||||
-- v:set("value")
|
||||
--
|
||||
-- channel.func_name(1,2,3):set("value")
|
||||
-- value = channel.func_name(1,2,3):get()
|
||||
--
|
||||
-- channel["func_name(1,2,3)"]:set("value")
|
||||
-- channel["func_name(1,2,3)"] = "value"
|
||||
-- value = channel["func_name(1,2,3)"]:get()
|
||||
--
|
||||
-- Note the use of the ':' operator to access the get() and set()
|
||||
-- methods.
|
||||
--
|
||||
-- Also notice the absence of the following constructs from the examples above:
|
||||
-- channel.func_name(1,2,3) = "value" -- this will NOT work
|
||||
-- value = channel.func_name(1,2,3) -- this will NOT work as expected
|
||||
--
|
||||
--
|
||||
-- Dialplan applications can be accessed through the global 'app' table.
|
||||
--
|
||||
-- app.Dial("DAHDI/1")
|
||||
-- app.dial("DAHDI/1")
|
||||
-- app["dial"]("DAHDI/1")
|
||||
--
|
||||
-- More examples can be found below.
|
||||
--
|
||||
-- An autoservice is automatically run while lua code is executing. The
|
||||
-- autoservice can be stopped and restarted using the autoservice_stop() and
|
||||
-- autoservice_start() functions. The autservice should be running before
|
||||
-- starting long running operations. The autoservice will automatically be
|
||||
-- stopped before executing applications and dialplan functions and will be
|
||||
-- restarted afterwards. The autoservice_status() function can be used to
|
||||
-- check the current status of the autoservice and will return true if an
|
||||
-- autoservice is currently running.
|
||||
--
|
||||
-- Note about naming conflicts:
|
||||
-- Lua allows you to refer to table entries using the '.' notation,
|
||||
-- I.E. app.goto(something), only if the entry doesn't conflict with an Lua
|
||||
-- reserved word. In the 'goto' example, with Lua 5.1 or earlier, 'goto' is
|
||||
-- not a reserved word so you'd be calling the Asterisk dialplan application
|
||||
-- 'goto'. Lua 5.2 however, introduced the 'goto' control statement which
|
||||
-- makes 'goto' a reserved word. This casues the interpreter to fail parsing
|
||||
-- the file and pbx_lua.so will fail to load. The same applies to any use of
|
||||
-- Lua tables including extensions, channels and any tables you create.
|
||||
--
|
||||
-- There are two ways around this: Since Lua is case-sensitive, you can use
|
||||
-- capitalized names, I.E. app.Goto(something) to refer to the Asterisk apps,
|
||||
-- functions, etc. Or you can use the full syntax, I.E. app["goto"](something).
|
||||
-- Both syntaxes are backwards compatible with earlier Lua versions. To make
|
||||
-- your Lua dialplans easier to maintain and to reduce the chance of future
|
||||
-- conflicts you may want to use the app["goto"](something) syntax for all
|
||||
-- table accesses.
|
||||
--
|
||||
|
||||
function outgoing_local(c, e)
|
||||
app.dial("DAHDI/1/" .. e, "", "")
|
||||
end
|
||||
|
||||
function demo_instruct()
|
||||
app.background("demo-instruct")
|
||||
app.waitexten()
|
||||
end
|
||||
|
||||
function demo_congrats()
|
||||
app.background("demo-congrats")
|
||||
demo_instruct()
|
||||
end
|
||||
|
||||
-- Answer the chanel and play the demo sound files
|
||||
function demo_start(context, exten)
|
||||
app.wait(1)
|
||||
app.answer()
|
||||
|
||||
channel.TIMEOUT("digit"):set(5)
|
||||
channel.TIMEOUT("response"):set(10)
|
||||
-- app.set("TIMEOUT(digit)=5")
|
||||
-- app.set("TIMEOUT(response)=10")
|
||||
|
||||
demo_congrats(context, exten)
|
||||
end
|
||||
|
||||
function demo_hangup()
|
||||
app.playback("demo-thanks")
|
||||
app.hangup()
|
||||
end
|
||||
|
||||
extensions = {
|
||||
demo = {
|
||||
s = demo_start;
|
||||
|
||||
["2"] = function()
|
||||
app.background("demo-moreinfo")
|
||||
demo_instruct()
|
||||
end;
|
||||
["3"] = function ()
|
||||
channel.LANGUAGE():set("fr") -- set the language to french
|
||||
demo_congrats()
|
||||
end;
|
||||
|
||||
["1000"] = function()
|
||||
-- See the naming conflict note above.
|
||||
app['goto']("default", "s", 1)
|
||||
end;
|
||||
|
||||
["1234"] = function()
|
||||
app.playback("transfer", "skip")
|
||||
-- do a dial here
|
||||
end;
|
||||
|
||||
["1235"] = function()
|
||||
app.voicemail("1234", "u")
|
||||
end;
|
||||
|
||||
["1236"] = function()
|
||||
app.dial("Console/dsp")
|
||||
app.voicemail(1234, "b")
|
||||
end;
|
||||
|
||||
["#"] = demo_hangup;
|
||||
t = demo_hangup;
|
||||
i = function()
|
||||
app.playback("invalid")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["500"] = function()
|
||||
app.playback("demo-abouttotry")
|
||||
app.dial("IAX2/guest@misery.digium.com/s@default")
|
||||
app.playback("demo-nogo")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["600"] = function()
|
||||
app.playback("demo-echotest")
|
||||
app.echo()
|
||||
app.playback("demo-echodone")
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
["8500"] = function()
|
||||
app.voicemailmain()
|
||||
demo_instruct()
|
||||
end;
|
||||
|
||||
};
|
||||
|
||||
default = {
|
||||
-- by default, do the demo
|
||||
include = {"demo"};
|
||||
};
|
||||
|
||||
public = {
|
||||
-- ATTENTION: If your Asterisk is connected to the internet and you do
|
||||
-- not have allowguest=no in sip.conf, everybody out there may use your
|
||||
-- public context without authentication. In that case you want to
|
||||
-- double check which services you offer to the world.
|
||||
--
|
||||
include = {"demo"};
|
||||
};
|
||||
|
||||
["local"] = {
|
||||
["_NXXXXXX"] = outgoing_local;
|
||||
};
|
||||
}
|
||||
|
||||
hints = {
|
||||
demo = {
|
||||
[1000] = "SIP/1000";
|
||||
[1001] = "SIP/1001";
|
||||
};
|
||||
|
||||
default = {
|
||||
["1234"] = "SIP/1234";
|
||||
};
|
||||
}
|
||||
156
asterisk/extensions_minivm.conf
Executable file
156
asterisk/extensions_minivm.conf
Executable file
@ -0,0 +1,156 @@
|
||||
; MINI-VOICEMAIL dialplan example
|
||||
; ---------------------------------------------------------------------------------------
|
||||
;
|
||||
; This is an example on how to use the Mini-Voicemail system to build
|
||||
; voicemail systems.
|
||||
;
|
||||
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
|
||||
; A macro to test the MINIVMACCOUNT dialplan function
|
||||
; Currently, accountcode and pincode is not used in the application
|
||||
; They where added to be used in dialplan scripting
|
||||
;
|
||||
;
|
||||
[macro-minivmfunctest]
|
||||
exten => s,1,set(account=${ARGV1})
|
||||
exten => minivm,n,verbose(1,-------------------- Minivm Function test - Accoutn ${account} -------------)
|
||||
exten => s,n,verbose(1,---- Has account: ${MINIVMACCOUNT(${account}:hasaccount)})
|
||||
exten => s,n,verbose(1,---- Fullname: ${MINIVMACCOUNT(${account}:fullname)})
|
||||
exten => s,n,verbose(1,---- Email: ${MINIVMACCOUNT(${account}:email)})
|
||||
exten => s,n,verbose(1,---- Pager: ${MINIVMACCOUNT(${account}:pager)})
|
||||
exten => s,n,verbose(1,---- E-mail template: ${MINIVMACCOUNT(${account}:etemplate)})
|
||||
exten => s,n,verbose(1,---- Pager template: ${MINIVMACCOUNT(${account}:ptemplate)})
|
||||
exten => s,n,verbose(1,---- Account code: ${MINIVMACCOUNT(${account}:accountcode)})
|
||||
exten => s,n,verbose(1,---- Path: ${MINIVMACCOUNT(${account}:path)})
|
||||
exten => s,n,verbose(1,---- Pincode: ${MINIVMACCOUNT(${account}:pincode)})
|
||||
exten => s,n,verbose(1,---- Time zone: ${MINIVMACCOUNT(${account}:timezone)})
|
||||
exten => s,n,verbose(1,---- Language: ${MINIVMACCOUNT(${account}:language)})
|
||||
; This requires setvar=customerclass=gold in the account configuration
|
||||
exten => s,n,verbose(1,---- Var:customerclass: ${MINIVMACCOUNT(${account}:customerclass)})
|
||||
|
||||
[minivm-scenario1]
|
||||
; minivmtest tests the dialplan function MINIVMACCOUNT
|
||||
; Check the output in the console with verbose set
|
||||
exten => minivmtest,1,answer
|
||||
exten => minivmtest,n,wait(0.5)
|
||||
exten => minivmtest,n,set(ACCOUNT=do-not-spam-me@example.com)
|
||||
exten => minivmtest,n,macro(minivmfunctest, ${ACCOUNT})
|
||||
exten => minivmtest,n,playback(beep)
|
||||
exten => minivmtest,n,hangup
|
||||
|
||||
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
|
||||
; "minivm" tests a full scenario
|
||||
; Remember that users may hangup
|
||||
; This works both for users with accounts in minivm.conf and by just giving an e-mail address
|
||||
; without configuring an account
|
||||
exten => minivm,1,answer
|
||||
exten => minivm,n,wait(0.5) ; Wait for Voip channels to settle
|
||||
exten => minivm,n,set(account=oej@example.com)
|
||||
exten => minivm,n,noop(------------------------------------------- Minivm Greet -------------)
|
||||
exten => minivm,n,minivmgreet(${account})
|
||||
exten => minivm,n,verbose(1,-- MINIVM_GREET_STATUS = ${MINIVM_GREET_STATUS} )
|
||||
exten => minivm,n,noop(------------------------------------------- Minivm Record -------------)
|
||||
exten => minivm,n,minivmRecord(${account},b)
|
||||
exten => minivm,n,goto(minivmcleanup,1)
|
||||
|
||||
; Cleanup after recording or hangup
|
||||
exten => minivmcleanup,1,noop(------------------------------------------- Minivm Notify -------------)
|
||||
;Increment voicemail counter with 1. The counter will be used in the e-mail message
|
||||
;and in the filename
|
||||
exten => minivmcleanup,n,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?minivmrecordfailure,1)
|
||||
exten => minivmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
|
||||
exten => minivmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
|
||||
exten => minivmcleanup,n,minivmNotify(${account})
|
||||
exten => minivmcleanup,n,verbose(1,-- MINIVM_NOTIFY_STATUS = ${MINIVM_NOTIFY_STATUS} )
|
||||
; Now, clean up after sending voicemail
|
||||
exten => minivmcleanup,n,noop(------------------------------------------- Minivm Delete -------------)
|
||||
exten => minivmcleanup,n,minivmdelete()
|
||||
exten => minivmcleanup,n,verbose(1,-- MINIVM_DELETE_STATUS = ${MINIVM_DELETE_STATUS} )
|
||||
|
||||
;Recording failed
|
||||
exten => minivmrecordfailure,1,playback(vm-sorry)
|
||||
exten => minivmrecordfailure,n,wait(1)
|
||||
exten => minivmrecordfailure,n,hangup
|
||||
|
||||
; If the user hangs up during the recording, we need to clean up
|
||||
; And send notifications
|
||||
exten => h,1,gotoif($[x${MINIVM_DELETE_STATUS} != x] ?h,stop)
|
||||
exten => h,n,noop(------------------------------------------- HANGUP during voicemail recording -------------)
|
||||
exten => h,n,goto(minivmcleanup,1)
|
||||
exten => h,n(stop),noop(---Minivm DONE----)
|
||||
|
||||
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
|
||||
; Extension to record a greeting message
|
||||
; Call this like:
|
||||
; macro(recordgreetings,alice@atlanta.example.com)
|
||||
;
|
||||
[macro-recordgreetings]
|
||||
exten => s,1,answer
|
||||
exten => s,n,wait(0.5)
|
||||
exten => s,n,set(account=${ARGV1])
|
||||
; This file give extra options not available here, needs to be edited
|
||||
; Change of password does not work
|
||||
exten => s,n(menu),background(vm-options)
|
||||
exten => 1,1,setvar(option=u)
|
||||
exten => 1,n,macro(minivmrec,${account},${option})
|
||||
exten => 1,n,goto(menu)
|
||||
exten => 2,1,setvar(option=b)
|
||||
exten => 2,n,macro(minivmrec,${account},${option})
|
||||
exten => 2,n,goto(menu)
|
||||
exten => 3,1,setvar(option=n)
|
||||
exten => 3,n,macro(minivmrec,${account},${option})
|
||||
exten => 3,n,goto(menu)
|
||||
exten => 4,1,setvar(option=t)
|
||||
exten => 4,n,macro(minivmrec,${account},${option})
|
||||
exten => 4,n,goto(menu)
|
||||
exten => *,1,playback(vm-thankyou)
|
||||
exten => *,n,wait(1)
|
||||
exten => *,n,hangup
|
||||
|
||||
exten => i,1,playback(invalid)
|
||||
exten => i,n,goto(menu)
|
||||
|
||||
[macro-minivmrec]
|
||||
exten => s,1,gotoif(${MINIVMACCOUNT(${account}:hasaccount)}?record)
|
||||
; Account is not configured in minivm.conf or realtime
|
||||
; Phony message, add something useful here
|
||||
exten => s,n,playback(privacy-incorrect)
|
||||
exten => s,n,macroreturn
|
||||
exten => record,1,minivmappmess(${ARGV1},${ARGV2})
|
||||
exten => record,n,noop(Recording status: ${MINIVM_APPMESS_STATUS})
|
||||
exten => record,n,macroreturn
|
||||
|
||||
;.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-
|
||||
; To set a counter and use a template for voicemail to users without acounts
|
||||
; use something like this
|
||||
;
|
||||
; email address is in the "account" channel variable. Set from ast_db or a script
|
||||
; based on called ID
|
||||
|
||||
exten => sendvoicemail,1,answer
|
||||
exten => sendvoicemail,n,wait(0.5)
|
||||
exten => sendvoicemail,n,set(domain=${CUT(account,@,2)})
|
||||
exten => sendvoicemail,n,set(country=${CUT(domain,.,2)})
|
||||
exten => sendvoicemail,n,minivmgreet(${account})
|
||||
exten => sendvoicemail,n,minivmRecord(${account},b)
|
||||
exten => sendvoicemail,n,goto(sendvmcleanup)
|
||||
|
||||
exten => sendvmcleanup,1,gotoif($[${MINIVM_RECORD_STATUS} != SUCCESS]?done)
|
||||
; The counter is set in the domain directory, so we don't create one directory per user
|
||||
; The counter has the email in the name of the counter, increase it
|
||||
; Set the MVM_COUNTER variable that we use in the template
|
||||
exten => sendvmcleanup,n,set(MINIVMCOUNTER(${account}:voicemailcounter:inc)=1)
|
||||
exten => sendvmcleanup,n,set(MVM_COUNTER = ${MINIVMCOUNTER(${account}:voicemailcounter)})
|
||||
; Increase a domain counter too, to see how many voicemails are sent to this domain
|
||||
; This is just for statistics
|
||||
exten => sendvmcleanup,n,set(MINIVMCOUNTER(${domain}:${domain}-all:inc) = 1)
|
||||
|
||||
; Send voicemail in e-mail with country-specific template
|
||||
; The template need to be defined in minivm.conf
|
||||
;
|
||||
exten => sendvmcleanup,n,minivmNotify(${account}, ${country}_email)
|
||||
exten => sendvmcleanup,n,minivmDelete()
|
||||
|
||||
exten => sendvmcleanup,n(done),wait(0.5)
|
||||
exten => sendvmcleanup,n,hangup
|
||||
|
||||
exten => h,1,gotoif($[${MINIVM_RECORD_STATUS} = SUCCESS]?sendvmcleanup,1))
|
||||
122
asterisk/features.conf
Executable file
122
asterisk/features.conf
Executable file
@ -0,0 +1,122 @@
|
||||
;
|
||||
; Sample Call Features (transfer, monitor/mixmonitor, etc) configuration
|
||||
;
|
||||
|
||||
; Note: From Asterisk 12 - All parking lot configuration is now done in res_parking.conf
|
||||
|
||||
[general]
|
||||
;transferdigittimeout => 3 ; Number of seconds to wait between digits when transferring a call
|
||||
; (default is 3 seconds)
|
||||
;xfersound = beep ; to indicate an attended transfer is complete
|
||||
;xferfailsound = beeperr ; to indicate a failed transfer
|
||||
;pickupexten = *8 ; Configure the pickup extension. (default is *8)
|
||||
;pickupsound = beep ; to indicate a successful pickup (default: no sound)
|
||||
;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound)
|
||||
;featuredigittimeout = 1000 ; Max time (ms) between digits for
|
||||
; feature activation (default is 1000 ms)
|
||||
;recordingfailsound = beeperr ; indicates that a one-touch monitor or one-touch mixmonitor feature failed
|
||||
; to be applied to the call. (default: no sound)
|
||||
;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds.
|
||||
;atxferdropcall = no ; If someone does an attended transfer, then hangs up before the transfer
|
||||
; target answers, then by default, the system will try to call back the
|
||||
; person that did the transfer. If this is set to "yes", the ringing
|
||||
; transfer target is immediately transferred to the transferee.
|
||||
;atxferloopdelay = 10 ; Number of seconds to sleep between retries (if atxferdropcall = no)
|
||||
;atxfercallbackretries = 2 ; Number of times to attempt to send the call back to the transferer.
|
||||
; By default, this is 2.
|
||||
;transferdialattempts = 3 ; Number of times that a transferer may attempt to dial an extension before
|
||||
; being kicked back to the original call.
|
||||
;transferretrysound = beep ; Sound to play when a transferer fails to dial a valid extension.
|
||||
;transferinvalidsound = beeperr ; Sound to play when a transferer fails to dial a valid extension and is out of retries.
|
||||
;atxferabort = *1 ; cancel the attended transfer
|
||||
;atxfercomplete = *2 ; complete the attended transfer, dropping out of the call
|
||||
;atxferthreeway = *3 ; complete the attended transfer, but stay in the call. This will turn the call into a multi-party bridge
|
||||
;atxferswap = *4 ; swap to the other party. Once an attended transfer has begun, this options may be used multiple times
|
||||
|
||||
; Note that the DTMF features listed below only work when two channels have answered and are bridged together.
|
||||
; They can not be used while the remote party is ringing or in progress. If you require this feature you can use
|
||||
; chan_local in combination with Answer to accomplish it.
|
||||
|
||||
[featuremap]
|
||||
;blindxfer => #1 ; Blind transfer (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
|
||||
;disconnect => *0 ; Disconnect (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
|
||||
;automon => *1 ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
|
||||
;atxfer => *2 ; Attended transfer -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
|
||||
;parkcall => #72 ; Park call (one step parking) -- Make sure to set the K and/or k option in the Dial() app call!
|
||||
;automixmon => *3 ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!
|
||||
|
||||
[applicationmap]
|
||||
; Note that the DYNAMIC_FEATURES channel variable must be set to use the features
|
||||
; defined here. The value of DYNAMIC_FEATURES should be the names of the features
|
||||
; to allow the channel to use separated by '#'. For example:
|
||||
;
|
||||
; Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3)
|
||||
;
|
||||
; (Note: The two leading underscores allow these feature settings to be set
|
||||
; on the outbound channels, as well. Otherwise, only the original channel
|
||||
; will have access to these features.)
|
||||
;
|
||||
; The syntax for declaring a dynamic feature is any of the following:
|
||||
;
|
||||
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]]
|
||||
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,"<AppArguments>"[,MOH_Class]]
|
||||
;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>([<AppArguments>])[,MOH_Class]
|
||||
|
||||
;
|
||||
; FeatureName -> This is the name of the feature used when setting the
|
||||
; DYNAMIC_FEATURES variable to enable usage of this feature.
|
||||
; DTMF_sequence -> This is the key sequence used to activate this feature.
|
||||
; ActivateOn -> This is the channel of the call that the application will be executed
|
||||
; on. Valid values are "self" and "peer". "self" means run the
|
||||
; application on the same channel that activated the feature. "peer"
|
||||
; means run the application on the opposite channel from the one that
|
||||
; has activated the feature.
|
||||
; ActivatedBy -> ActivatedBy is no longer honored. The feature is activated by which
|
||||
; channel DYNAMIC_FEATURES includes the feature is on. Use a pre-dial
|
||||
; handler to set different values for DYNAMIC_FEATURES on the channels.
|
||||
; Historic values are: "caller", "callee", and "both".
|
||||
; Application -> This is the application to execute.
|
||||
; AppArguments -> These are the arguments to be passed into the application. If you need
|
||||
; commas in your arguments, you should use either the second or third
|
||||
; syntax, above.
|
||||
; MOH_Class -> This is the music on hold class to play while the idle
|
||||
; channel waits for the feature to complete. If left blank,
|
||||
; no music will be played.
|
||||
;
|
||||
|
||||
;
|
||||
; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk
|
||||
; applications. When applications are used in extensions.conf, they are executed
|
||||
; by the PBX core. In this case, these applications are executed outside of the
|
||||
; PBX core, so it does *not* make sense to use any application which has any
|
||||
; concept of dialplan flow. Examples of this would be things like Goto,
|
||||
; Background, WaitExten, and many more. The exceptions to this are Gosub and
|
||||
; Macro routines which must complete for the call to continue.
|
||||
;
|
||||
; Enabling these features means that the PBX needs to stay in the media flow and
|
||||
; media will not be re-directed if DTMF is sent in the media stream.
|
||||
;
|
||||
; Example Usage:
|
||||
;
|
||||
;testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play
|
||||
; ;tt-monkeys to the opposite channel
|
||||
;
|
||||
; Set arbitrary channel variables, based upon CALLERID number (Note that the application
|
||||
; argument contains commas)
|
||||
;retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})
|
||||
;
|
||||
;pauseMonitor => #1,self/callee,Pausemonitor ;Allow the callee to pause monitoring
|
||||
; ;on their channel
|
||||
;unpauseMonitor => #3,self/callee,UnPauseMonitor ;Allow the callee to unpause monitoring
|
||||
; ;on their channel
|
||||
|
||||
; Dynamic Feature Groups:
|
||||
; Dynamic feature groups are groupings of features defined in [applicationmap]
|
||||
; that can have their own custom key mappings. To give a channel access to a dynamic
|
||||
; feature group, add the group name to the value of the DYNAMIC_FEATURES variable.
|
||||
;
|
||||
; example:
|
||||
; [myGroupName] ; defines the group named myGroupName
|
||||
; testfeature => #9 ; associates testfeature with the group and the keycode '#9'.
|
||||
; pauseMonitor => ; associates pauseMonitor with the group and uses the keycode specified
|
||||
; ; in the [applicationmap].
|
||||
34
asterisk/festival.conf
Executable file
34
asterisk/festival.conf
Executable file
@ -0,0 +1,34 @@
|
||||
;
|
||||
; Festival Configuration
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; Host which runs the festival server (default : localhost);
|
||||
;
|
||||
;host=localhost
|
||||
;
|
||||
; Port on host where the festival server runs (default : 1314)
|
||||
;
|
||||
;port=1314
|
||||
;
|
||||
; Use cache (yes, no - defaults to no)
|
||||
;
|
||||
;usecache=yes
|
||||
;
|
||||
; If usecache=yes, a directory to store waveform cache files.
|
||||
; The cache is never cleared (yet), so you must take care of cleaning it
|
||||
; yourself (just delete any or all files from the cache).
|
||||
; THIS DIRECTORY *MUST* EXIST and must be writable from the asterisk process.
|
||||
; Defaults to /tmp/
|
||||
;
|
||||
;cachedir=/var/lib/asterisk/festivalcache/
|
||||
;
|
||||
; Festival command to send to the server.
|
||||
; Defaults to: (tts_textasterisk "%s" 'file)(quit)\n
|
||||
; %s is replaced by the desired text to say. The command MUST end with a
|
||||
; (quit) directive, or the cache handling mechanism will hang. Do not
|
||||
; forget the \n at the end.
|
||||
;
|
||||
;festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n
|
||||
;
|
||||
;
|
||||
107
asterisk/followme.conf
Executable file
107
asterisk/followme.conf
Executable file
@ -0,0 +1,107 @@
|
||||
; Find-Me / Follow-Me Configuration File
|
||||
[general]
|
||||
;
|
||||
featuredigittimeout=>5000
|
||||
; The number of ms to wait for a digit input for the callee on whether to take the call or
|
||||
; not before we consider them "done" entering digits.
|
||||
;
|
||||
enable_callee_prompt=>true
|
||||
; Enable prompting a callee to either accept or reject the forwarded call.
|
||||
; If disabled, the optional prompting for caller name (option 'a') is
|
||||
; likewise disabled. Enabled by default.
|
||||
;
|
||||
takecall=>1
|
||||
; The global default keypress for the callee to take taking the current call. This can be
|
||||
; a single digit or multiple digits. Default is "1".
|
||||
; Note this string must not be a substring of declinecall.
|
||||
;
|
||||
declinecall=>2
|
||||
; The global default keypress for the callee to decline taking the current call. This can
|
||||
; be a single digit or multiple digits. Default is "2".
|
||||
; Note this string must not be a substring of takecall.
|
||||
;
|
||||
call_from_prompt=>followme/call-from
|
||||
; The global default for the 'Incoming call from' message.
|
||||
;
|
||||
norecording_prompt=>followme/no-recording
|
||||
; The global default for the 'You have an incoming call' message when the caller elects
|
||||
; not to leave their name or the option isn't set for them to do so.
|
||||
;
|
||||
options_prompt=>followme/options
|
||||
; The global default for the 'Press 1 to accept this call or press 2 to decline it' message.
|
||||
;
|
||||
pls_hold_prompt=>followme/pls-hold-while-try
|
||||
; The global default for 'Please hold while we try and connect your call' message.
|
||||
;
|
||||
status_prompt=>followme/status
|
||||
; The global default for 'The party you're calling isn't at their desk' message.
|
||||
;
|
||||
sorry_prompt=>followme/sorry
|
||||
; The global default for 'I'm sorry, but we were unable to locate your party' message.
|
||||
;
|
||||
connecting_prompt=>
|
||||
; The global default sound file name for 'Please say hello to the caller' message.
|
||||
; Setting to an empty string skips playing the prompt. The default is no prompt
|
||||
; file name.
|
||||
|
||||
[default]
|
||||
musicclass=>default
|
||||
; The moh class that should be used for the caller while they are waiting to be connected.
|
||||
context=>default
|
||||
; The context to dial the numbers from
|
||||
number=>01233456,25
|
||||
; The a follow-me number to call. The format is:
|
||||
; number=> <number to call[&2nd #[&...]]>[,<timeout value in seconds>[,<order in follow-me>]]
|
||||
; You can specify as many of these numbers as you like. They will be dialed in the
|
||||
; order that you specify them in the config file OR as specified with the order field
|
||||
; on the number prompt. As you can see from the example, forked dialing of multiple
|
||||
; numbers in the same step is supported with this application if you'd like to dial
|
||||
; multiple numbers in the same followme step.
|
||||
;
|
||||
; The timeout value is the amount of time allowed between the time the dialing step
|
||||
; starts and the callee answers. The callee then has until the timeout of the last
|
||||
; step to make a choice on whether to take the call or not. That being the case,
|
||||
; you may want to make the timeout on the last step longer to give enough time to
|
||||
; make the choice to accept or not.
|
||||
enable_callee_prompt=>true
|
||||
; Enable prompting the callee to accept the forwarded call. The default
|
||||
; is the global value.
|
||||
takecall=>1
|
||||
; The keypress for the callee to take taking the current call. This can be
|
||||
; a single digit or multiple digits. Default is the global default.
|
||||
; Note this string must not be a substring of declinecall.
|
||||
;
|
||||
declinecall=>2
|
||||
; The keypress for the callee to decline taking the current call. This can
|
||||
; be a single digit or multiple digits. Default is the global default.
|
||||
; Note this string must not be a substring of takecall.
|
||||
;
|
||||
call_from_prompt=>followme/call-from
|
||||
; The 'Incoming call from' message prompt. Default is the global default.
|
||||
;
|
||||
norecording_prompt=>followme/no-recording
|
||||
; The 'You have an incoming call' message prompt when the caller elects
|
||||
; not to leave their name or the option isn't set for them to do so. Default
|
||||
; is the global default.
|
||||
;
|
||||
options_prompt=>followme/options
|
||||
; The 'Press 1 to accept this call or press 2 to decline it' message prompt.
|
||||
; Default is the global default.
|
||||
;
|
||||
pls_hold_prompt=>followme/pls-hold-while-try
|
||||
; The 'Please hold while we try and connect your call' message prompt.
|
||||
; Default is the global default.
|
||||
;
|
||||
status_prompt=>followme/status
|
||||
; The 'The party you're calling isn't at their desk' message prompt.
|
||||
; Default is the global default.
|
||||
;
|
||||
sorry_prompt=>followme/sorry
|
||||
; The 'I'm sorry, but we were unable to locate your party' message prompt. Default
|
||||
; is the global default.
|
||||
;
|
||||
connecting_prompt=>
|
||||
; The sound file name for 'Please say hello to the caller' message. Default is the
|
||||
; global default.
|
||||
;
|
||||
|
||||
112
asterisk/func_odbc.conf
Executable file
112
asterisk/func_odbc.conf
Executable file
@ -0,0 +1,112 @@
|
||||
;
|
||||
; func_odbc.conf
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; Asterisk uses separate connections for every database operation.
|
||||
; If single_db_connection is enabled then func_odbc will use a single
|
||||
; database connection per DSN.
|
||||
; This option exists for those who expect that a second func_odbc call
|
||||
; works on the same connection. That allows you to do a LAST_INSERT_ID()
|
||||
; in a second func_odbc call.
|
||||
; Note that you'll need additional dialplan locks for this behaviour to work.
|
||||
; There are better ways: using stored procedures/functions instead.
|
||||
; This option is disabled by default.
|
||||
;single_db_connection=no
|
||||
;
|
||||
;
|
||||
; Each context is a separately defined function. By convention, all
|
||||
; functions are entirely uppercase, so the defined contexts should also
|
||||
; be all-uppercase, but there is nothing that enforces this. All functions
|
||||
; are case-sensitive, however.
|
||||
;
|
||||
; For substitution, you have ${ARG1}, ${ARG2} ... ${ARGn}
|
||||
; for the arguments to each SQL statement.
|
||||
;
|
||||
; In addition, for write statements, you have ${VAL1}, ${VAL2} ... ${VALn}
|
||||
; parsed, just like arguments, for the values. In addition, if you want the
|
||||
; whole value, never mind the parsing, you can get that with ${VALUE}.
|
||||
;
|
||||
;
|
||||
; If you have data which may potentially contain single ticks, you may wish
|
||||
; to use the dialplan function SQL_ESC() to escape the data prior to its
|
||||
; inclusion in the SQL statement.
|
||||
;
|
||||
;
|
||||
; The following options are available in this configuration file:
|
||||
;
|
||||
; dsn An alias for "writehandle."
|
||||
; readhandle A comma-separated list of DSNs (from res_odbc.conf) to use when
|
||||
; executing the readsql statement. Each DSN is tried, in
|
||||
; succession, until the statement succeeds. You may specify up to
|
||||
; 5 DSNs per function class. If not specified, it will default to
|
||||
; the value of "writehandle" or "dsn," if specified.
|
||||
; writehandle A comma-separated list of DSNs (from res_odbc.conf) to use when
|
||||
; executing the writesql statement. The same rules apply as to
|
||||
; readhandle.
|
||||
; readsql The statement to execute when reading from the function class.
|
||||
; writesql The statement to execute when writing to the function class.
|
||||
; insertsql The statement to execute when writing to the function class
|
||||
; succeeds, but initially indicates that 0 rows were affected.
|
||||
; prefix Normally, all function classes are prefixed with "ODBC" to keep
|
||||
; them uniquely named. You may choose to change this prefix, which
|
||||
; may be useful to segregate a collection of certain function
|
||||
; classes from others.
|
||||
; escapecommas This option may be used to turn off the default behavior of
|
||||
; escaping commas which occur within a field. If commas are
|
||||
; escaped (the default behavior), then fields containing commas
|
||||
; will be treated as a single value when assigning to ARRAY() or
|
||||
; HASH(). If commas are not escaped, then values will be separated
|
||||
; at the comma within fields. Please note that turning this option
|
||||
; off is incompatible with the functionality of HASH().
|
||||
; synopsis Appears in the synopsis field for the command
|
||||
; 'core show function <function name>'
|
||||
; syntax Appears in the syntax field for the command
|
||||
; 'core show function <function name>'
|
||||
; mode This option may be set to 'multirow' to allow the function
|
||||
; specified to return more than a single row. However, this
|
||||
; changes the way that func_odbc normally works. Instead of the
|
||||
; invocation of the function returning a row, it returns an opaque
|
||||
; ID, which may be passed to ODBC_FETCH() to return each row in
|
||||
; turn. ODBC_FETCH_STATUS returns SUCCESS or FAILURE, to indicate
|
||||
; whether any results were stored, and you should call ODBC_Finish
|
||||
; on the ID to clean up any remaining results when you are done
|
||||
; with the query. Also, the variable ODBCROWS is set initially,
|
||||
; which may be used in an iterative fashion to return each row in
|
||||
; the result.
|
||||
; Please note that multirow queries are isolated to the channel,
|
||||
; and rows may not be fetched outside of the channel where the
|
||||
; query was initially performed. Additionally, as the results are
|
||||
; associated with a channel, mode=multirow is incompatible with
|
||||
; the global space.
|
||||
; rowlimit Rowlimit limits the total number of rows which can be stored for
|
||||
; that query. For mode=multirow, otherwise, func_odbc will
|
||||
; attempt to store all rows in the resultset, up to the maximum
|
||||
; amount of memory. In normal mode, rowlimit can be set to allow
|
||||
; additional rows to be fetched, rather than just the first one.
|
||||
; These additional rows can be returned by using the name of the
|
||||
; function which was called to retrieve the first row as an
|
||||
; argument to ODBC_FETCH().
|
||||
|
||||
|
||||
; ODBC_SQL - Allow an SQL statement to be built entirely in the dialplan
|
||||
[SQL]
|
||||
writehandle=mysql1
|
||||
readsql=${ARG1}
|
||||
|
||||
; ODBC_ANTISOLICIT - A blacklist for known solicitors.
|
||||
[ANTISOLICIT]
|
||||
dsn=mysql1,mysql2 ; Use mysql1 as the primary handle, but fall back to mysql2
|
||||
; if mysql1 is down. Supports up to 5 comma-separated
|
||||
; DSNs. "dsn" may also be specified as "readhandle" and
|
||||
; "writehandle", if it is important to separate reads and
|
||||
; writes to different databases.
|
||||
readsql=SELECT COUNT(*) FROM known_solicitors WHERE callerid='${SQL_ESC(${ARG1})}'
|
||||
syntax=<callerid>
|
||||
synopsis=Check if a specified callerid is contained in the known solicitors database
|
||||
|
||||
; ODBC_PRESENCE - Retrieve and update presence
|
||||
[PRESENCE]
|
||||
writehandle=mysql1
|
||||
readsql=SELECT location FROM presence WHERE id='${SQL_ESC(${ARG1})}'
|
||||
writesql=UPDATE presence SET location='${SQL_ESC(${VAL1})}' WHERE id='${SQL_ESC(${ARG1})}'
|
||||
32
asterisk/hep.conf
Executable file
32
asterisk/hep.conf
Executable file
@ -0,0 +1,32 @@
|
||||
;
|
||||
; res_hep Module configuration for Asterisk
|
||||
;
|
||||
|
||||
;
|
||||
; Note that this configuration file is consumed by res_hep, which is responsible
|
||||
; for the HEPv3 protocol manipulation and managing the connection to the Homer
|
||||
; capture server. Additional modules provide specific messages to be sent to
|
||||
; the Homer server:
|
||||
; - res_hep_pjsip: Send SIP messages transmitted/received by the PJSIP stack
|
||||
; - res_hep_rtcp: Send RTCP information (all channels)
|
||||
;
|
||||
|
||||
; All settings are currently set in the general section.
|
||||
[general]
|
||||
enabled = no ; Enable/disable forwarding of packets to a
|
||||
; HEP server. Default is "yes".
|
||||
capture_address = 192.168.1.1:9061 ; The address of the HEP capture server.
|
||||
capture_password = foo ; If specified, the authorization passsword
|
||||
; for the HEP server. If not specified, no
|
||||
; authorization password will be sent.
|
||||
capture_id = 1234 ; A unique integer identifier for this
|
||||
; server. This ID will be embedded sent
|
||||
; with each packet from this server.
|
||||
uuid_type = call-id ; Specify the preferred source for the Homer
|
||||
; correlation UUID. Valid options are:
|
||||
; - 'call-id' for the PJSIP or chan_sip SIP
|
||||
; Call-ID
|
||||
; - 'channel' for the Asterisk channel name
|
||||
; Note: If 'call-id' is specified but the
|
||||
; channel is not PJSIP or chan_sip then the
|
||||
; Asterisk channel name will be used instead.
|
||||
8
asterisk/http.conf
Executable file
8
asterisk/http.conf
Executable file
@ -0,0 +1,8 @@
|
||||
[general]
|
||||
enabled = yes
|
||||
bindaddr = 0.0.0.0
|
||||
bindport=8080
|
||||
tlsenable=yes
|
||||
tlsbindaddr=0.0.0.0:8089
|
||||
tlscertfile=/etc/asterisk/keys/cert.pem
|
||||
tlsprivatekey=/etc/asterisk/keys/privkey.pem
|
||||
132
asterisk/http.conf.default
Executable file
132
asterisk/http.conf.default
Executable file
@ -0,0 +1,132 @@
|
||||
;
|
||||
; Asterisk Built-in mini-HTTP server
|
||||
;
|
||||
;
|
||||
; Note about Asterisk documentation:
|
||||
; If Asterisk was installed from a tarball, then the HTML documentation should
|
||||
; be installed in the static-http/docs directory which is
|
||||
; (/var/lib/asterisk/static-http/docs) on linux by default. If the Asterisk
|
||||
; HTTP server is enabled in this file by setting the "enabled", "bindaddr",
|
||||
; and "bindport" options, then you should be able to view the documentation
|
||||
; remotely by browsing to:
|
||||
; http://<server_ip>:<bindport>/static/docs/index.html
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; The name of the server, advertised in both the Server field in HTTP
|
||||
; response message headers, as well as the <address /> element in certain HTTP
|
||||
; response message bodies. If not furnished here, "Asterisk/{version}" will be
|
||||
; used as a default value for the Server header field and the <address />
|
||||
; element. Setting this property to a blank value will result in the omission
|
||||
; of the Server header field from HTTP response message headers and the
|
||||
; <address /> element from HTTP response message bodies.
|
||||
;
|
||||
servername=Asterisk
|
||||
;
|
||||
; Whether HTTP/HTTPS interface is enabled or not. Default is no.
|
||||
; This also affects manager/rawman/mxml access (see manager.conf)
|
||||
;
|
||||
;enabled=yes
|
||||
;
|
||||
; Address to bind to, both for HTTP and HTTPS. You MUST specify
|
||||
; a bindaddr in order for the HTTP server to run. There is no
|
||||
; default value.
|
||||
;
|
||||
bindaddr=127.0.0.1
|
||||
;
|
||||
; Port to bind to for HTTP sessions (default is 8088)
|
||||
;
|
||||
;bindport=8088
|
||||
;
|
||||
; Prefix allows you to specify a prefix for all requests
|
||||
; to the server. The default is blank. If uncommented
|
||||
; all requests must begin with /asterisk
|
||||
;
|
||||
;prefix=asterisk
|
||||
;
|
||||
; sessionlimit specifies the maximum number of httpsessions that will be
|
||||
; allowed to exist at any given time. (default: 100)
|
||||
;
|
||||
;sessionlimit=100
|
||||
;
|
||||
; session_inactivity specifies the number of milliseconds to wait for
|
||||
; more data over the HTTP connection before closing it.
|
||||
;
|
||||
; Default: 30000
|
||||
;session_inactivity=30000
|
||||
;
|
||||
; session_keep_alive specifies the number of milliseconds to wait for
|
||||
; the next HTTP request over a persistent connection.
|
||||
;
|
||||
; Set to 0 to disable persistent HTTP connections.
|
||||
; Default: 15000
|
||||
;session_keep_alive=15000
|
||||
;
|
||||
; Whether Asterisk should serve static content from static-http
|
||||
; Default is no.
|
||||
;
|
||||
;enable_static=yes
|
||||
;
|
||||
; Whether Asterisk should serve a status page showing the running
|
||||
; configuration of this built-in HTTP server.
|
||||
; Default is yes.
|
||||
;
|
||||
;enable_status=no
|
||||
;
|
||||
; Redirect one URI to another. This is how you would set a
|
||||
; default page.
|
||||
; Syntax: redirect=<from here> <to there>
|
||||
; For example, if you are using the Asterisk-gui,
|
||||
; it is convenient to enable the following redirect:
|
||||
;
|
||||
;redirect = / /static/config/index.html
|
||||
;
|
||||
; HTTPS support. In addition to enabled=yes, you need to
|
||||
; explicitly enable tls, define the port to use,
|
||||
; and have a certificate somewhere.
|
||||
;tlsenable=yes ; enable tls - default no.
|
||||
;tlsbindaddr=0.0.0.0:8089 ; address and port to bind to - default is bindaddr and port 8089.
|
||||
;
|
||||
;tlscertfile=</path/to/certificate.pem> ; path to the certificate file (*.pem) only.
|
||||
;tlsprivatekey=</path/to/private.pem> ; path to private key file (*.pem) only.
|
||||
; If no path is given for tlscertfile or tlsprivatekey, default is to look in current
|
||||
; directory. If no tlsprivatekey is given, default is to search tlscertfile for private key.
|
||||
;
|
||||
; To produce a certificate you can e.g. use openssl. This places both the cert and
|
||||
; private in same .pem file.
|
||||
; openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
|
||||
;
|
||||
; tlscipher= ; The list of allowed ciphers
|
||||
; ; if none are specified the following cipher
|
||||
; ; list will be used instead:
|
||||
; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
|
||||
; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
|
||||
; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
|
||||
; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
|
||||
; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
|
||||
; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
|
||||
; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
|
||||
; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
|
||||
; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
|
||||
;
|
||||
; tlsdisablev1=yes ; Disable TLSv1 support - if not set this defaults to "yes"
|
||||
; tlsdisablev11=yes ; Disable TLSv1.1 support - if not set this defaults to "no"
|
||||
; tlsdisablev12=yes ; Disable TLSv1.2 support - if not set this defaults to "no"
|
||||
;
|
||||
; tlsservercipherorder=yes ; Use the server preference order instead of the client order
|
||||
; ; Defaults to "yes"
|
||||
;
|
||||
; The post_mappings section maps URLs to real paths on the filesystem. If a
|
||||
; POST is done from within an authenticated manager session to one of the
|
||||
; configured POST mappings, then any files in the POST will be placed in the
|
||||
; configured directory.
|
||||
;
|
||||
;[post_mappings]
|
||||
;
|
||||
; NOTE: You need a valid HTTP AMI mansession_id cookie with the manager
|
||||
; config permission to POST files.
|
||||
;
|
||||
; In this example, if the prefix option is set to "asterisk", then using the
|
||||
; POST URL: /asterisk/uploads will put files in /var/lib/asterisk/uploads/.
|
||||
;uploads = /var/lib/asterisk/uploads/
|
||||
;
|
||||
669
asterisk/iax.conf
Executable file
669
asterisk/iax.conf
Executable file
@ -0,0 +1,669 @@
|
||||
;
|
||||
; Inter-Asterisk eXchange v2 (IAX2) Channel Driver configuration
|
||||
;
|
||||
; This configuration is read when the chan_iax2.so module is loaded, and is
|
||||
; re-read when the module is reloaded, such as when invoking the CLI command:
|
||||
;
|
||||
; *CLI> iax2 reload
|
||||
;
|
||||
|
||||
; General settings, like port number to bind to, and an option address (the
|
||||
; default is to bind to all local addresses).
|
||||
|
||||
[general]
|
||||
|
||||
; Listener Addresses
|
||||
;
|
||||
; Use the 'bindaddr' and 'bindport' options to specify on which address and port
|
||||
; the IAX2 channel driver will listen for incoming requests.
|
||||
;
|
||||
;
|
||||
|
||||
;bindport=4569 ; The default port to listen on
|
||||
; NOTE: bindport must be specified BEFORE bindaddr or
|
||||
; may be specified on a specific bindaddr if followed by
|
||||
; colon and port (e.g. bindaddr=192.168.0.1:4569) or for
|
||||
; IPv6 the address needs to be in brackets then colon
|
||||
; and port (e.g. bindaddr=[2001:db8::1]:4569).
|
||||
|
||||
;bindaddr=192.168.0.1 ; You can specify 'bindaddr' more than once to bind to
|
||||
; multiple addresses, but the first will be the
|
||||
; default. IPv6 addresses are accepted.
|
||||
|
||||
;
|
||||
; Set 'iaxcompat' to yes if you plan to use layered switches or some other
|
||||
; scenario which may cause some delay when doing a lookup in the dialplan. It
|
||||
; incurs a small performance hit to enable it. This option causes Asterisk to
|
||||
; spawn a separate thread when it receives an IAX2 DPREQ (Dialplan Request)
|
||||
; instead of blocking while it waits for a response.
|
||||
;
|
||||
; Accepted values: yes, no
|
||||
; Default value: no
|
||||
;
|
||||
;iaxcompat=yes
|
||||
;
|
||||
|
||||
;
|
||||
; Disable UDP checksums (if nochecksums is set, then no checkums will
|
||||
; be calculated/checked on systems supporting this feature)
|
||||
;
|
||||
; Accepted values: yes, no
|
||||
; Default value: no
|
||||
;
|
||||
;nochecksums=yes
|
||||
;
|
||||
|
||||
;
|
||||
; For increased security against brute force password attacks enable
|
||||
; 'delayreject' which will delay the sending of authentication reject for REGREQ
|
||||
; or AUTHREP if there is a password.
|
||||
;
|
||||
; Accepted values: yes, no
|
||||
; Default value: no
|
||||
;
|
||||
;delayreject=yes
|
||||
;
|
||||
|
||||
;
|
||||
; You may specify a global default AMA flag for iaxtel calls. These flags are
|
||||
; used in the generation of call detail records.
|
||||
;
|
||||
; Accepted values: default, omit, billing, documentation
|
||||
; Default value: default
|
||||
;
|
||||
;amaflags=billing
|
||||
;
|
||||
|
||||
;
|
||||
; ADSI (Analog Display Services Interface) can be enabled if you have (or may
|
||||
; have) ADSI compatible CPE equipment.
|
||||
;
|
||||
; Accepted values: yes, no
|
||||
; Default value: no
|
||||
;
|
||||
;adsi=yes
|
||||
;
|
||||
|
||||
;
|
||||
; Whether or not to perform an SRV lookup on outbound calls.
|
||||
;
|
||||
; Accepted values: yes, no
|
||||
; Default value: no
|
||||
;
|
||||
;srvlookup=yes
|
||||
;
|
||||
|
||||
;
|
||||
; You may specify a default account for Call Detail Records (CDRs) in addition to
|
||||
; specifying on a per-user basis.
|
||||
;
|
||||
; Accepted values: Any string value up to 19 characters in length
|
||||
; Default value: <empty>
|
||||
;
|
||||
;accountcode=lss0101
|
||||
;
|
||||
|
||||
;
|
||||
; You may specify a global default language for users. This can be specified
|
||||
; also on a per-user basis. If omitted, will fallback to English (en).
|
||||
;
|
||||
; Accepted values: A language tag such as 'en' or 'es'
|
||||
; Default value: en
|
||||
;
|
||||
;language=en
|
||||
;
|
||||
|
||||
;
|
||||
; This option specifies a preference for which music-on-hold class this channel
|
||||
; should listen to when put on hold if the music class has not been set on the
|
||||
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
|
||||
; channel putting this one on hold did not suggest a music class.
|
||||
;
|
||||
; If this option is set to "passthrough", then the hold message will always be
|
||||
; passed through as signalling instead of generating hold music locally.
|
||||
;
|
||||
; This option may be specified globally, or on a per-user or per-peer basis.
|
||||
;
|
||||
; Accepted values: passthrough, or any music-on-hold class name
|
||||
; Default value: <empty>
|
||||
;
|
||||
;mohinterpret=default
|
||||
;
|
||||
|
||||
;
|
||||
; The 'mohsuggest' option specifies which music on hold class to suggest to the
|
||||
; peer channel when this channel places the peer on hold. It may be specified
|
||||
; globally or on a per-user or per-peer basis.
|
||||
;
|
||||
;mohsuggest=default
|
||||
;
|
||||
|
||||
;
|
||||
; Specify bandwidth of low, medium, or high to control which codecs are used
|
||||
; in general.
|
||||
;
|
||||
bandwidth=low
|
||||
;
|
||||
|
||||
;
|
||||
; You can also fine tune codecs here using "allow" and "disallow" clauses with
|
||||
; specific codecs. Use "all" to represent all formats.
|
||||
;
|
||||
;allow=all
|
||||
;disallow=g723.1
|
||||
disallow=lpc10
|
||||
;allow=gsm
|
||||
;
|
||||
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; Jitter Buffer
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
|
||||
;
|
||||
; You can adjust several parameters relating to the jitter buffer. The jitter
|
||||
; buffer's function is to compensate for varying network delay.
|
||||
;
|
||||
; All of the jitter buffer settings are in milliseconds. The jitter buffer
|
||||
; works for INCOMING audio only - the outbound audio will be dejittered by the
|
||||
; jitter buffer at the other end.
|
||||
;
|
||||
; jitterbuffer=yes|no: global default as to whether you want
|
||||
; the jitter buffer at all.
|
||||
;
|
||||
; maxjitterbuffer: a maximum size for the jitter buffer.
|
||||
; Setting a reasonable maximum here will prevent the call delay
|
||||
; from rising to silly values in extreme situations; you'll hear
|
||||
; SOMETHING, even though it will be jittery.
|
||||
;
|
||||
; resyncthreshold: when the jitterbuffer notices a significant change in delay
|
||||
; that continues over a few frames, it will resync, assuming that the change in
|
||||
; delay was caused by a timestamping mix-up. The threshold for noticing a
|
||||
; change in delay is measured as twice the measured jitter plus this resync
|
||||
; threshold.
|
||||
; Resyncing can be disabled by setting this parameter to -1.
|
||||
;
|
||||
; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
|
||||
; should return in a row. Since some clients do not send CNG/DTX frames to
|
||||
; indicate silence, the jitterbuffer will assume silence has begun after
|
||||
; returning this many interpolations. This prevents interpolating throughout
|
||||
; a long silence.
|
||||
;
|
||||
; jittertargetextra: number of milliseconds by which the new jitter buffer
|
||||
; will pad its size. the default is 40, so without modification, the new
|
||||
; jitter buffer will set its size to the jitter value plus 40 milliseconds.
|
||||
; increasing this value may help if your network normally has low jitter,
|
||||
; but occasionally has spikes.
|
||||
;
|
||||
|
||||
jitterbuffer=no
|
||||
;maxjitterbuffer=1000
|
||||
;maxjitterinterps=10
|
||||
;resyncthreshold=1000
|
||||
;jittertargetextra=40
|
||||
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
; IAX2 Encryption
|
||||
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
|
||||
|
||||
;
|
||||
; Enable IAX2 encryption. The default is no.
|
||||
;
|
||||
;encryption=yes
|
||||
;
|
||||
|
||||
;
|
||||
; Force encryption insures no connection is established unless both sides
|
||||
; support encryption. By turning this option on, encryption is automatically
|
||||
; turned on as well. The default is no.
|
||||
;
|
||||
;forceencryption=yes
|
||||
;
|
||||
|
||||
; This option defines the maximum payload in bytes an IAX2 trunk can support at
|
||||
; a given time. The best way to explain this is to provide an example. If the
|
||||
; maximum number of calls to be supported is 800, and each call transmits 20ms
|
||||
; frames of audio using ulaw:
|
||||
;
|
||||
; (8000hz / 1000ms) * 20ms * 1 byte per sample = 160 bytes per frame
|
||||
;
|
||||
; The maximum load in bytes is:
|
||||
;
|
||||
; (160 bytes per frame) * (800 calls) = 128000 bytes
|
||||
;
|
||||
; Once this limit is reached, calls may be dropped or begin to lose audio.
|
||||
; Depending on the codec in use and number of channels to be supported this value
|
||||
; may need to be raised, but in most cases the default value is large enough.
|
||||
;
|
||||
; trunkmaxsize = 128000 ; defaults to 128000 bytes, which supports up to 800
|
||||
; calls of ulaw at 20ms a frame.
|
||||
|
||||
; With a large amount of traffic on IAX2 trunks, there is a risk of bad voice
|
||||
; quality when allowing the Linux system to handle fragmentation of UDP packets.
|
||||
; Depending on the size of each payload, allowing the OS to handle fragmentation
|
||||
; may not be very efficient. This setting sets the maximum transmission unit for
|
||||
; IAX2 UDP trunking. The default is 1240 bytes which means if a trunk's payload
|
||||
; is over 1240 bytes for every 20ms it will be broken into multiple 1240 byte
|
||||
; messages. Zero disables this functionality and let's the OS handle
|
||||
; fragmentation.
|
||||
;
|
||||
; trunkmtu = 1240 ; trunk data will be sent in 1240 byte messages.
|
||||
|
||||
; trunkfreq sets how frequently trunk messages are sent in milliseconds. This
|
||||
; value is 20ms by default, which means the trunk will send all the data queued
|
||||
; to it in the past 20ms. By increasing the time between sending trunk messages,
|
||||
; the trunk's payload size will increase as well. Note, depending on the size
|
||||
; set by trunkmtu, messages may be sent more often than specified. For example
|
||||
; if a trunk's message size grows to the trunkmtu size before 20ms is reached
|
||||
; that message will be sent immediately. Acceptable values are between 10ms and
|
||||
; 1000ms.
|
||||
;
|
||||
; trunkfreq=20 ; How frequently to send trunk msgs (in ms). This is 20ms by
|
||||
; default.
|
||||
|
||||
; Should we send timestamps for the individual sub-frames within trunk frames?
|
||||
; There is a small bandwidth use for these (less than 1kbps/call), but they
|
||||
; ensure that frame timestamps get sent end-to-end properly. If both ends of
|
||||
; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
|
||||
; length for your codecs, you can probably suppress these. The receiver must
|
||||
; also support this feature, although they do not also need to have it enabled.
|
||||
;
|
||||
; trunktimestamps=yes
|
||||
|
||||
; Minimum and maximum amounts of time that IAX2 peers can request as a
|
||||
; registration expiration interval (in seconds).
|
||||
; minregexpire = 60
|
||||
; maxregexpire = 60
|
||||
|
||||
; IAX2 helper threads
|
||||
|
||||
; Establishes the number of iax helper threads to handle I/O.
|
||||
; iaxthreadcount = 10
|
||||
|
||||
; Establishes the number of extra dynamic threads that may be spawned to handle I/O
|
||||
; iaxmaxthreadcount = 100
|
||||
|
||||
;
|
||||
; We can register with another IAX2 server to let him know where we are
|
||||
; in case we have a dynamic IP address for example
|
||||
;
|
||||
; Register with tormenta using username marko and password secretpass
|
||||
;
|
||||
;register => marko:secretpass@tormenta.linux-support.net
|
||||
;
|
||||
; Register joe at remote host with no password
|
||||
;
|
||||
;register => joe@remotehost:5656
|
||||
;
|
||||
; Register marko at tormenta.linux-support.net using RSA key "torkey"
|
||||
;
|
||||
;register => marko:[torkey]@tormenta.linux-support.net
|
||||
;
|
||||
; Sample Registration for iaxtel
|
||||
;
|
||||
; Visit http://www.iaxtel.com to register with iaxtel. Replace "user"
|
||||
; and "pass" with your username and password for iaxtel. Incoming
|
||||
; calls arrive at the "s" extension of "default" context.
|
||||
;
|
||||
;register => user:pass@iaxtel.com
|
||||
;
|
||||
; Sample Registration for IAX2 + FWD
|
||||
;
|
||||
; To register using IAX2 with FWD, it must be enabled by visiting the URL
|
||||
; http://www.fwdnet.net/index.php?section_id=112
|
||||
;
|
||||
; Note that you need an extension in you default context which matches
|
||||
; your free world dialup number. Please replace "FWDNumber" with your
|
||||
; FWD number and "passwd" with your password.
|
||||
;
|
||||
;register => FWDNumber:passwd@iax.fwdnet.net
|
||||
;
|
||||
; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the
|
||||
; perceived external network address has changed. When the stun_monitor is installed and
|
||||
; configured, chan_iax will renew all outbound registrations when the monitor detects any sort
|
||||
; of network change has occurred. By default this option is enabled, but only takes effect once
|
||||
; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not
|
||||
; generate all outbound registrations on a network change, use the option below to disable
|
||||
; this feature.
|
||||
;
|
||||
; subscribe_network_change_event = yes ; on by default
|
||||
;
|
||||
; You can enable authentication debugging to increase the amount of
|
||||
; debugging traffic.
|
||||
;
|
||||
;authdebug = yes
|
||||
;
|
||||
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
|
||||
;tos=ef
|
||||
;cos=5
|
||||
;
|
||||
; If regcontext is specified, Asterisk will dynamically create and destroy
|
||||
; a NoOp priority 1 extension for a given peer who registers or unregisters
|
||||
; with us. The actual extension is the 'regexten' parameter of the registering
|
||||
; peer or its name if 'regexten' is not provided. More than one regexten
|
||||
; may be supplied if they are separated by '&'. Patterns may be used in
|
||||
; regexten.
|
||||
;
|
||||
;regcontext=iaxregistrations
|
||||
;
|
||||
; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
|
||||
; then we cancel the whole thing (that's enough time for one retransmission
|
||||
; only). This is used to keep things from stalling for a long time for a host
|
||||
; that is not available, but would be ill advised for bad connections. In
|
||||
; addition to 'yes' or 'no' you can also specify a number of milliseconds.
|
||||
; See 'qualify' for individual peers to turn on for just a specific peer.
|
||||
;
|
||||
autokill=yes
|
||||
;
|
||||
; codecpriority controls the codec negotiation of an inbound IAX2 call.
|
||||
; This option is inherited to all user entities. It can also be defined
|
||||
; in each user entity separately which will override the setting in general.
|
||||
;
|
||||
; The valid values are:
|
||||
;
|
||||
; caller - Consider the callers preferred order ahead of the host's.
|
||||
; host - Consider the host's preferred order ahead of the caller's.
|
||||
; disabled - Disable the consideration of codec preference altogether.
|
||||
; (this is the original behaviour before preferences were added)
|
||||
; reqonly - Same as disabled, only do not consider capabilities if
|
||||
; the requested format is not available the call will only
|
||||
; be accepted if the requested format is available.
|
||||
;
|
||||
; The default value is 'host'
|
||||
;
|
||||
;codecpriority=host
|
||||
;
|
||||
; allowfwdownload controls whether this host will serve out firmware to
|
||||
; IAX2 clients which request it. This has only been used for the IAXy,
|
||||
; and it has been recently proven that this firmware distribution method
|
||||
; can be used as a source of traffic amplification attacks. Also, the
|
||||
; IAXy firmware has not been updated for at least 18 months, so unless
|
||||
; you are provisioning IAXys in a secure network, we recommend that you
|
||||
; leave this option to the default, off.
|
||||
;
|
||||
;allowfwdownload=yes
|
||||
|
||||
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
|
||||
; just like friends added from the config file only on a
|
||||
; as-needed basis? (yes|no)
|
||||
|
||||
;rtsavesysname=yes ; Save systemname in realtime database at registration
|
||||
; Default = no
|
||||
|
||||
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
|
||||
; If set to yes, when a IAX2 peer registers successfully,
|
||||
; the IP address, the origination port, the registration period,
|
||||
; and the username of the peer will be set to database via realtime.
|
||||
; If not present, defaults to 'yes'.
|
||||
|
||||
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
|
||||
; as if it had just registered? (yes|no|<seconds>)
|
||||
; If set to yes, when the registration expires, the friend will
|
||||
; vanish from the configuration until requested again.
|
||||
; If set to an integer, friends expire within this number of
|
||||
; seconds instead of the registration interval.
|
||||
|
||||
;rtignoreregexpire=yes ; When reading a peer from Realtime, if the peer's registration
|
||||
; has expired based on its registration interval, used the stored
|
||||
; address information regardless. (yes|no)
|
||||
|
||||
;parkinglot=edvina ; Default parkinglot for IAX2 peers and users
|
||||
; This can also be configured per device
|
||||
; Parkinglots are defined in features.conf
|
||||
|
||||
;
|
||||
; The following two options are used to disable call token validation for the
|
||||
; purposes of interoperability with IAX2 endpoints that do not yet support it.
|
||||
;
|
||||
; Call token validation can be set as optional for a single IP address or IP
|
||||
; address range by using the 'calltokenoptional' option. 'calltokenoptional' is
|
||||
; only a global option.
|
||||
;
|
||||
;calltokenoptional=209.16.236.73/255.255.255.0
|
||||
;
|
||||
; By setting 'requirecalltoken=no', call token validation becomes optional for
|
||||
; that peer/user. By setting 'requirecalltoken=auto', call token validation
|
||||
; is optional until a call token supporting peer registers successfully using
|
||||
; call token validation. This is used as an indication that from now on, we
|
||||
; can require it from this peer. So, requirecalltoken is internally set to yes.
|
||||
; requirecalltoken may only be used in peer/user/friend definitions,
|
||||
; not in the global scope.
|
||||
; By default, 'requirecalltoken=yes'.
|
||||
;
|
||||
;requirecalltoken=no
|
||||
;
|
||||
; Maximum time allowed for call token authentication handshaking. Default is 10 seconds.
|
||||
; Use higher values in lagged or high packet loss networks.
|
||||
;
|
||||
;calltokenexpiration=10
|
||||
|
||||
;
|
||||
; These options are used to limit the amount of call numbers allocated to a
|
||||
; single IP address. Before changing any of these values, it is highly encouraged
|
||||
; to read the user guide associated with these options first. In most cases, the
|
||||
; default values for these options are sufficient.
|
||||
;
|
||||
; The 'maxcallnumbers' option limits the amount of call numbers allowed for each
|
||||
; individual remote IP address. Once an IP address reaches it's call number
|
||||
; limit, no more new connections are allowed until the previous ones close. This
|
||||
; option can be used in a peer definition as well, but only takes effect for
|
||||
; the IP of a dynamic peer after it completes registration.
|
||||
;
|
||||
;maxcallnumbers=512
|
||||
;
|
||||
; The 'maxcallnumbers_nonvalidated' is used to set the combined number of call
|
||||
; numbers that can be allocated for connections where call token validation
|
||||
; has been disabled. Unlike the 'maxcallnumbers' option, this limit is not
|
||||
; separate for each individual IP address. Any connection resulting in a
|
||||
; non-call token validated call number being allocated contributes to this
|
||||
; limit. For use cases, see the call token user guide. This option's
|
||||
; default value of 8192 should be sufficient in most cases.
|
||||
;
|
||||
;maxcallnumbers_nonvalidated=1024
|
||||
;
|
||||
; The [callnumberlimits] section allows custom call number limits to be set
|
||||
; for specific IP addresses and IP address ranges. These limits take precedence
|
||||
; over the global 'maxcallnumbers' option, but may still be overridden by a
|
||||
; peer defined 'maxcallnumbers' entry. Note that these limits take effect
|
||||
; for every individual address within the range, not the range as a whole.
|
||||
;
|
||||
;[callnumberlimits]
|
||||
;10.1.1.0/255.255.255.0 = 24
|
||||
;10.1.2.0/255.255.255.0 = 32
|
||||
;
|
||||
|
||||
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
|
||||
; in square brackets. For example, the Caller*ID value 555.5555 becomes 5555555
|
||||
; when this option is enabled. Disabling this option results in no modification
|
||||
; of the Caller*ID value, which is necessary when the Caller*ID represents something
|
||||
; that must be preserved. This option can only be used in the [general] section.
|
||||
; By default this option is on.
|
||||
;
|
||||
;shrinkcallerid=yes ; on by default
|
||||
|
||||
; Guest sections for unauthenticated connection attempts. Just specify an
|
||||
; empty secret, or provide no secret section.
|
||||
;
|
||||
[guest]
|
||||
type=user
|
||||
context=public
|
||||
callerid="Guest IAX User"
|
||||
|
||||
;
|
||||
; Trust Caller*ID Coming from iaxtel.com
|
||||
;
|
||||
[iaxtel]
|
||||
type=user
|
||||
context=default
|
||||
auth=rsa
|
||||
inkeys=iaxtel
|
||||
|
||||
;
|
||||
; Trust Caller*ID Coming from iax.fwdnet.net
|
||||
;
|
||||
[iaxfwd]
|
||||
type=user
|
||||
context=default
|
||||
auth=rsa
|
||||
inkeys=freeworlddialup
|
||||
|
||||
;
|
||||
; Trust Caller*ID delivered over DUNDi/e164
|
||||
;
|
||||
;[dundi]
|
||||
;type=user
|
||||
;dbsecret=dundi/secret
|
||||
;context=dundi-e164-local
|
||||
|
||||
;
|
||||
; Further user sections may be added, specifying a context and a secret used
|
||||
; for connections with that given authentication name. Limited IP based
|
||||
; access control is allowed by use of "permit", "deny", and "acl" keywords.
|
||||
; Multiple rules are permitted. Multiple permitted contexts may be specified,
|
||||
; in which case the first will be the default. You can also override
|
||||
; Caller*ID so that when you receive a call you set the Caller*ID to be what
|
||||
; you want instead of trusting what the remote user provides
|
||||
;
|
||||
; There are three authentication methods that are supported: md5, plaintext,
|
||||
; and rsa. The least secure is "plaintext", which sends passwords cleartext
|
||||
; across the net. "md5" uses a challenge/response md5 sum arrangement, but
|
||||
; still requires both ends have plain text access to the secret. "rsa" allows
|
||||
; unidirectional secret knowledge through public/private keys. If "rsa"
|
||||
; authentication is used, "inkeys" is a list of acceptable public keys on the
|
||||
; local system that can be used to authenticate the remote peer, separated by
|
||||
; the ":" character. "outkey" is a single, private key to use to authenticate
|
||||
; to the other side. Public keys are named /var/lib/asterisk/keys/<name>.pub
|
||||
; while private keys are named /var/lib/asterisk/keys/<name>.key. Private
|
||||
; keys should always be 3DES encrypted.
|
||||
;
|
||||
;
|
||||
; NOTE: All hostnames and IP addresses in this file are for example purposes
|
||||
; only; you should not expect any of them to actually be available for
|
||||
; your use.
|
||||
;
|
||||
;[markster]
|
||||
;type=user
|
||||
;context=default
|
||||
;context=local
|
||||
;auth=md5,plaintext,rsa
|
||||
;secret=markpasswd
|
||||
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
|
||||
; cause the given audio file to
|
||||
; be played upon completion of
|
||||
; an attended transfer to the
|
||||
; target of the transfer.
|
||||
;dbsecret=mysecrets/place ; Secrets can be stored in astdb, too
|
||||
;transfer=no ; Disable IAX2 native transfer
|
||||
;transfer=mediaonly ; When doing IAX2 native transfers, transfer only
|
||||
; the media stream
|
||||
;jitterbuffer=yes ; Override the global setting and enable the jitter
|
||||
; buffer for this user
|
||||
;maxauthreq=10 ; Set the maximum number of outstanding AUTHREQs
|
||||
; waiting for replies. If this limit is reached,
|
||||
; any further authentication will be blocked, until
|
||||
; the pending requests expire or a reply is
|
||||
; received.
|
||||
;callerid="Mark Spencer" <(256) 428-6275>
|
||||
;deny=0.0.0.0/0.0.0.0
|
||||
;accountcode=markster0101
|
||||
;permit=209.16.236.73/255.255.255.0
|
||||
;language=en ; Use english as default language
|
||||
;encryption=yes ; Enable IAX2 encryption. The default is no.
|
||||
;keyrotate=off ; This is a compatibility option for older versions
|
||||
; of IAX2 that do not support key rotation with
|
||||
; encryption. This option will disable the
|
||||
; IAX_COMMAND_RTENC message. The default is on.
|
||||
|
||||
;
|
||||
; Peers may also be specified, with a secret and a remote hostname.
|
||||
;
|
||||
;[demo]
|
||||
;type=peer
|
||||
;username=asterisk
|
||||
;secret=supersecret
|
||||
;host=192.168.10.10
|
||||
;description=My IAX2 Peer ; Description of this peer, as listed by
|
||||
; 'iax2 show peers'
|
||||
;sendani=no
|
||||
;host=asterisk.linux-support.net
|
||||
;port=5036
|
||||
;mask=255.255.255.255
|
||||
;qualify=yes ; Make sure this peer is alive.
|
||||
;qualifysmoothing = yes ; Use an average of the last two PONG results to
|
||||
; reduce falsely detected LAGGED hosts. The default
|
||||
; is 'no.'
|
||||
;qualifyfreqok = 60000 ; How frequently to ping the peer when everything
|
||||
; seems to be OK, in milliseconds.
|
||||
;qualifyfreqnotok = 10000 ; How frequently to ping the peer when it's either
|
||||
; LAGGED or UNAVAILABLE, in milliseconds.
|
||||
;jitterbuffer=no ; Turn off jitter buffer for this peer
|
||||
;
|
||||
;encryption=yes ; Enable IAX2 encryption. The default is no.
|
||||
;keyrotate=off ; This is a compatibility option for older versions
|
||||
; of IAX2 that do not support key rotation with
|
||||
; encryption. This option will disable the
|
||||
; IAX_COMMAND_RTENC message. The default is 'on.'
|
||||
|
||||
; Peers can remotely register as well, so that they can be mobile. Default
|
||||
; IPs can also optionally be given but are not required. Caller*ID can be
|
||||
; suggested to the other side as well if it is for example a phone instead of
|
||||
; another PBX.
|
||||
;connectedline=yes ; Set if connected line and redirecting information updates
|
||||
; ; are passed between Asterisk servers for this peer.
|
||||
; ; yes - Sending and receiving updates are enabled.
|
||||
; ; send - Only send updates.
|
||||
; ; receive - Only process received updates.
|
||||
; ; no - Sending and receiving updates are disabled.
|
||||
; ; Default is "no".
|
||||
; ;
|
||||
; ; Note: Because of an incompatibility between Asterisk v1.4
|
||||
; ; and Asterisk v1.8 or later, this option must be set
|
||||
; ; to "no" toward the Asterisk v1.4 peer. A symptom of the
|
||||
; ; incompatibility is the call gets disconnected unexpectedly.
|
||||
|
||||
|
||||
;[dynamichost]
|
||||
;host=dynamic
|
||||
;secret=mysecret
|
||||
; Note: app_voicemail mailboxes must be in the form of mailbox@context.
|
||||
;mailbox=1234 ; Notify about mailbox 1234
|
||||
;inkeys=key1:key2
|
||||
;peercontext=local ; Default context to request for calls to peer
|
||||
;defaultip=216.207.245.34
|
||||
;callerid="Some Host" <(256) 428-6011>
|
||||
|
||||
;[biggateway]
|
||||
;type=peer
|
||||
;host=192.168.0.1
|
||||
;description=Gateway to PSTN
|
||||
;context=*
|
||||
;secret=myscret
|
||||
;trunk=yes ; Use IAX2 trunking with this host
|
||||
;timezone=America/New_York ; Set a timezone for the date/time IE
|
||||
|
||||
;
|
||||
; Friends are a shortcut for creating a user and a peer with the same values.
|
||||
;
|
||||
|
||||
;[marko]
|
||||
;type=friend
|
||||
;host=dynamic
|
||||
;regexten=1234
|
||||
;secret=moofoo ; Multiple secrets may be specified. For a "user", all
|
||||
;secret=foomoo ; specified entries will be accepted as valid. For a "peer",
|
||||
;secret=shazbot ; only the last specified secret will be used.
|
||||
;context=default
|
||||
;permit=0.0.0.0/0.0.0.0
|
||||
;acl=example_named_acl
|
||||
|
||||
;
|
||||
; With immediate=yes, an IAX2 phone or a phone on an IAXy acts as a hot-line
|
||||
; which goes immediately to the s extension when picked up. Useful for
|
||||
; elevator phones, manual service, or other similar applications.
|
||||
;
|
||||
;[manual]
|
||||
;type=friend
|
||||
;host=dynamic
|
||||
;immediate=yes ; go immediately to s extension when picked up
|
||||
;secret=moofoo ; when immediate=yes is specified, secret is required
|
||||
;context=number-please ; we start at the s extension in this context
|
||||
;
|
||||
80
asterisk/iaxprov.conf
Executable file
80
asterisk/iaxprov.conf
Executable file
@ -0,0 +1,80 @@
|
||||
;
|
||||
; IAX2 Provisioning Information
|
||||
;
|
||||
; Contains provisioning information for templates and for specific service
|
||||
; entries.
|
||||
;
|
||||
; Templates provide a group of settings from which provisioning takes place.
|
||||
; A template may be based upon any template that has been specified before
|
||||
; it. If the template that an entry is based on is not specified then it is
|
||||
; presumed to be 'default' (unless it is the first of course).
|
||||
;
|
||||
; Templates which begin with 'si-' are used for provisioning units with
|
||||
; specific service identifiers. For example the entry "si-000364000126"
|
||||
; would be used when the device with the corresponding service identifier of
|
||||
; "000364000126" attempts to register or make a call.
|
||||
;
|
||||
[default]
|
||||
;
|
||||
; The port number the device should use to bind to. The default is 4569.
|
||||
;
|
||||
;port=4569
|
||||
;
|
||||
; server is our PRIMARY server for registration and placing calls
|
||||
;
|
||||
;server=192.168.69.3
|
||||
;
|
||||
; altserver is the BACKUP server for registration and placing calls in the
|
||||
; event the primary server is unavailable.
|
||||
;
|
||||
;altserver=192.168.69.4
|
||||
;
|
||||
; port is the port number to use for IAX2 outbound. The connections to the
|
||||
; server and altserver -- default is of course 4569.
|
||||
;serverport=4569
|
||||
;
|
||||
; language is the preferred language for the device
|
||||
;
|
||||
;language=en
|
||||
;
|
||||
; codec is the requested codec. The iaxy supports ulaw and adpcm
|
||||
;
|
||||
codec=ulaw
|
||||
;
|
||||
; flags is a comma separated list of flags which the device should
|
||||
; use and may contain any of the following keywords:
|
||||
;
|
||||
; "register" - Register with server
|
||||
; "secure" - Do not accept calls / provisioning not originated by the server
|
||||
; "heartbeat" - Generate status packets on port 9999 sent to 255.255.255.255
|
||||
; "debug" - Output extra debugging to port 9999
|
||||
;
|
||||
; Note that use can use += and -= to adjust parameters
|
||||
;
|
||||
flags=register,heartbeat
|
||||
;
|
||||
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of this parameter.
|
||||
;tos=ef
|
||||
;
|
||||
; Example iaxy provisioning
|
||||
;
|
||||
;[si-000364000126]
|
||||
;user=iaxy
|
||||
;pass=bitsy
|
||||
;flags += debug
|
||||
|
||||
;[si-000364000127]
|
||||
;user=iaxy2
|
||||
;pass=bitsy2
|
||||
;template=si-000364000126
|
||||
;flags += debug
|
||||
|
||||
;
|
||||
;[*]
|
||||
;
|
||||
; If specified, the '*' provisioning is used for all devices which do not
|
||||
; have another provisioning entry within the file. If unspecified, no
|
||||
; provisioning will take place for devices which have no entry. DO NOT
|
||||
; USE A '*' PROVISIONING ENTRY UNLESS YOU KNOW WHAT YOU'RE DOING.
|
||||
;
|
||||
;template=default
|
||||
747
asterisk/indications.conf
Executable file
747
asterisk/indications.conf
Executable file
@ -0,0 +1,747 @@
|
||||
;
|
||||
; indications.conf
|
||||
;
|
||||
; Configuration file for location specific tone indications
|
||||
;
|
||||
|
||||
;
|
||||
; NOTE:
|
||||
; When adding countries to this file, please keep them in alphabetical
|
||||
; order according to the 2-character country codes!
|
||||
;
|
||||
; The [general] category is for certain global variables.
|
||||
; All other categories are interpreted as location specific indications
|
||||
;
|
||||
|
||||
[general]
|
||||
country=us ; default location
|
||||
|
||||
|
||||
; [example]
|
||||
; description = string
|
||||
; The full name of your country, in English.
|
||||
; ringcadence = num[,num]*
|
||||
; List of durations the physical bell rings.
|
||||
; dial = tonelist
|
||||
; Set of tones to be played when one picks up the hook.
|
||||
; busy = tonelist
|
||||
; Set of tones played when the receiving end is busy.
|
||||
; congestion = tonelist
|
||||
; Set of tones played when there is some congestion (on the network?)
|
||||
; callwaiting = tonelist
|
||||
; Set of tones played when there is a call waiting in the background.
|
||||
; dialrecall = tonelist
|
||||
; Not well defined; many phone systems play a recall dial tone after hook
|
||||
; flash.
|
||||
; record = tonelist
|
||||
; Set of tones played when call recording is in progress.
|
||||
; info = tonelist
|
||||
; Set of tones played with special information messages (e.g., "number is
|
||||
; out of service")
|
||||
; 'name' = tonelist
|
||||
; Every other variable will be available as a shortcut for the "PlayList" command
|
||||
; but will not be used automatically by Asterisk.
|
||||
;
|
||||
;
|
||||
; The tonelist itself is defined by a comma-separated sequence of elements.
|
||||
; Each element consist of a frequency (f) with an optional duration (in ms)
|
||||
; attached to it (f/duration). The frequency component may be a mixture of two
|
||||
; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
|
||||
; The implicit modulation depth is fixed at 90%, though.
|
||||
; If the list element starts with a !, that element is NOT repeated,
|
||||
; therefore, only if all elements start with !, the tonelist is time-limited,
|
||||
; all others will repeat indefinitely.
|
||||
;
|
||||
; concisely:
|
||||
; element = [!]freq[+|*freq2][/duration]
|
||||
; tonelist = element[,element]*
|
||||
;
|
||||
|
||||
[at]
|
||||
description = Austria
|
||||
ringcadence = 1000,5000
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
dial = 420
|
||||
busy = 420/400,0/400
|
||||
ring = 420/1000,0/5000
|
||||
congestion = 420/200,0/200
|
||||
callwaiting = 420/40,0/1960
|
||||
dialrecall = 420
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/80,0/14920
|
||||
info = 950/330,1450/330,1850/330,0/1000
|
||||
stutter = 380+420
|
||||
|
||||
[au]
|
||||
description = Australia
|
||||
; Reference http://www.acif.org.au/__data/page/3303/S002_2001.pdf
|
||||
; Normal Ring
|
||||
ringcadence = 400,200,400,2000
|
||||
; Distinctive Ring 1 - Forwarded Calls
|
||||
; 400,400,200,200,400,1400
|
||||
; Distinctive Ring 2 - Selective Ring 2 + Operator + Recall
|
||||
; 400,400,200,2000
|
||||
; Distinctive Ring 3 - Multiple Subscriber Number 1
|
||||
; 200,200,400,2200
|
||||
; Distinctive Ring 4 - Selective Ring 1 + Centrex
|
||||
; 400,2600
|
||||
; Distinctive Ring 5 - Selective Ring 3
|
||||
; 400,400,200,400,200,1400
|
||||
; Distinctive Ring 6 - Multiple Subscriber Number 2
|
||||
; 200,400,200,200,400,1600
|
||||
; Distinctive Ring 7 - Multiple Subscriber Number 3 + Data Privacy
|
||||
; 200,400,200,400,200,1600
|
||||
; Tones
|
||||
dial = 413+438
|
||||
busy = 425/375,0/375
|
||||
ring = 413+438/400,0/200,413+438/400,0/2000
|
||||
; XXX Congestion: Should reduce by 10 db every other cadence XXX
|
||||
congestion = 425/375,0/375,420/375,0/375
|
||||
callwaiting = 425/200,0/200,425/200,0/4400
|
||||
dialrecall = 413+438
|
||||
; Record tone used for Call Intrusion/Recording or Conference
|
||||
record = !425/1000,!0/15000,425/360,0/15000
|
||||
info = 425/2500,0/500
|
||||
; Other Australian Tones
|
||||
; The STD "pips" indicate the call is not an untimed local call
|
||||
std = !525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100,!0/100,!525/100
|
||||
; Facility confirmation tone (eg. Call Forward Activated)
|
||||
facility = 425
|
||||
; Message Waiting "stutter" dialtone
|
||||
stutter = 413+438/100,0/40
|
||||
; Ringtone for calls to Telstra mobiles
|
||||
ringmobile = 400+450/400,0/200,400+450/400,0/2000
|
||||
|
||||
[bg]
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
description = Bulgaria
|
||||
ringcadence = 1000,4000
|
||||
;
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 425/150,0/150,425/150,0/4000
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
record = 1400/425,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
stutter = 425/1500,0/100
|
||||
|
||||
[br]
|
||||
description = Brazil
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/250,0/250
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250,425/750,0/250
|
||||
callwaiting = 425/50,0/1000
|
||||
; Dialrecall not used in Brazil standard (using UK standard)
|
||||
dialrecall = 350+440
|
||||
; Record tone is not used in Brazil, use busy tone
|
||||
record = 425/250,0/250
|
||||
; Info not used in Brazil standard (using UK standard)
|
||||
info = 950/330,1400/330,1800/330
|
||||
stutter = 350+440
|
||||
|
||||
[be]
|
||||
description = Belgium
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,3000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/3000
|
||||
congestion = 425/167,0/167
|
||||
callwaiting = 1400/175,0/175,1400/175,0/3500
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
info = 900/330,1400/330,1800/330,0/1000
|
||||
stutter = 425/1000,0/250
|
||||
|
||||
[ch]
|
||||
description = Switzerland
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/200,0/200,425/200,0/4000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/80,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
stutter = 425+340/1100,0/1100
|
||||
|
||||
[cl]
|
||||
description = Chile
|
||||
; According to specs from Telefonica CTC Chile
|
||||
ringcadence = 1000,3000
|
||||
dial = 400
|
||||
busy = 400/500,0/500
|
||||
ring = 400/1000,0/3000
|
||||
congestion = 400/200,0/200
|
||||
callwaiting = 400/250,0/8750
|
||||
dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
|
||||
record = 1400/500,0/15000
|
||||
info = 950/333,1400/333,1800/333,0/1000
|
||||
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
|
||||
|
||||
[cn]
|
||||
description = China
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 450
|
||||
busy = 450/350,0/350
|
||||
ring = 450/1000,0/4000
|
||||
congestion = 450/700,0/700
|
||||
callwaiting = 450/400,0/4000
|
||||
dialrecall = 450
|
||||
record = 950/400,0/10000
|
||||
info = 450/100,0/100,450/100,0/100,450/100,0/100,450/400,0/400
|
||||
; STUTTER - not specified
|
||||
stutter = 450+425
|
||||
|
||||
[cz]
|
||||
description = Czech Republic
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425/330,0/330,425/660,0/660
|
||||
busy = 425/330,0/330
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/165,0/165
|
||||
callwaiting = 425/330,0/9000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425/330,0/330,425/660,0/660
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/14000
|
||||
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
|
||||
; STUTTER - not specified
|
||||
stutter = 425/450,0/50
|
||||
|
||||
[de]
|
||||
description = Germany
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/480,0/480
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/240,0/240
|
||||
callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/80,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
stutter = 425+400
|
||||
|
||||
[dk]
|
||||
description = Denmark
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = !425/200,!0/600,!425/200,!0/3000,!425/200,!0/200,!425/200,0
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/80,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
; STUTTER - not specified
|
||||
stutter = 425/450,0/50
|
||||
|
||||
[ee]
|
||||
description = Estonia
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/300,0/300
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
; CALLWAIT not in accordance to ITU
|
||||
callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = 425/650,0/25
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
; INFO not in accordance to ITU
|
||||
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
|
||||
; STUTTER not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[es]
|
||||
description = Spain
|
||||
ringcadence = 1500,3000
|
||||
dial = 425
|
||||
busy = 425/200,0/200
|
||||
ring = 425/1500,0/3000
|
||||
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
|
||||
callwaiting = 425/175,0/175,425/175,0/3500
|
||||
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
|
||||
record = 1400/500,0/15000
|
||||
info = 950/330,0/1000
|
||||
dialout = 500
|
||||
; STUTTER not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
|
||||
[fi]
|
||||
description = Finland
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/300,0/300
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/150,0/150,425/150,0/8000
|
||||
dialrecall = 425/650,0/25
|
||||
record = 1400/500,0/15000
|
||||
info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
|
||||
stutter = 425/650,0/25
|
||||
|
||||
[fr]
|
||||
description = France
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1500,3500
|
||||
; Dialtone can also be 440+330
|
||||
dial = 440
|
||||
busy = 440/500,0/500
|
||||
ring = 440/1500,0/3500
|
||||
; CONGESTION - not specified
|
||||
congestion = 440/250,0/250
|
||||
callwait = 440/300,0/10000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330
|
||||
stutter = !440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,!440/100,!0/100,440
|
||||
|
||||
[gr]
|
||||
description = Greece
|
||||
ringcadence = 1000,4000
|
||||
dial = 425/200,0/300,425/700,0/800
|
||||
busy = 425/300,0/300
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/150,0/150,425/150,0/8000
|
||||
dialrecall = 425/650,0/25
|
||||
record = 1400/400,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
|
||||
stutter = 425/650,0/25
|
||||
|
||||
[hu]
|
||||
description = Hungary
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1250,3750
|
||||
dial = 425
|
||||
busy = 425/300,0/300
|
||||
ring = 425/1250,0/3750
|
||||
congestion = 425/300,0/300
|
||||
callwaiting = 425/40,0/1960
|
||||
dialrecall = 425+450
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/400,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
|
||||
stutter = 350+375+400
|
||||
|
||||
[id]
|
||||
description = Indonesia
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 425/150,0/150,425/150,0/10000
|
||||
info = !950/330,!1400/330,!1800/330,0/1000
|
||||
|
||||
[il]
|
||||
description = Israel
|
||||
ringcadence = 1000,3000
|
||||
dial = 414
|
||||
busy = 414/500,0/500
|
||||
ring = 414/1000,0/3000
|
||||
congestion = 414/250,0/250
|
||||
callwaiting = 414/100,0/100,414/100,0/100,414/600,0/3000
|
||||
dialrecall = !414/100,!0/100,!414/100,!0/100,!414/100,!0/100,414
|
||||
record = 1400/500,0/15000
|
||||
info = 1000/330,1400/330,1800/330,0/1000
|
||||
stutter = !414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,!414/160,!0/160,414
|
||||
|
||||
|
||||
[in]
|
||||
description = India
|
||||
ringcadence = 400,200,400,2000
|
||||
dial = 400*25
|
||||
busy = 400/750,0/750
|
||||
ring = 400*25/400,0/200,400*25/400,0/2000
|
||||
congestion = 400/250,0/250
|
||||
callwaiting = 400/200,0/100,400/200,0/7500
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,0/1000
|
||||
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,400*25
|
||||
|
||||
[it]
|
||||
description = Italy
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
dial = 425/200,0/200,425/600,0/1000
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/400,0/100,425/250,0/100,425/150,0/14000
|
||||
dialrecall = 470/400,425/400
|
||||
record = 1400/400,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
|
||||
stutter = 470/400,425/400
|
||||
|
||||
[lt]
|
||||
description = Lithuania
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/350,0/350
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/150,0/150,425/150,0/4000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = 425/500,0/50
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
|
||||
; STUTTER - not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[jp]
|
||||
description = Japan
|
||||
ringcadence = 1000,2000
|
||||
dial = 400
|
||||
busy = 400/500,0/500
|
||||
ring = 400+15/1000,0/2000
|
||||
congestion = 400/500,0/500
|
||||
callwaiting = 400+16/500,0/8000
|
||||
dialrecall = !400/200,!0/200,!400/200,!0/200,!400/200,!0/200,400
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,0
|
||||
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
|
||||
|
||||
[mx]
|
||||
description = Mexico
|
||||
ringcadence = 2000,4000
|
||||
dial = 425
|
||||
busy = 425/250,0/250
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 425/200,0/600,425/200,0/10000
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
record = 1400/500,0/15000
|
||||
info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
|
||||
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
|
||||
|
||||
[my]
|
||||
description = Malaysia
|
||||
ringcadence = 2000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/400,0/200,425/400,0/2000
|
||||
congestion = 425/500,0/500
|
||||
; STUTTER - not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[nl]
|
||||
description = Netherlands
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
ringcadence = 1000,4000
|
||||
; Most of these 425's can also be 450's
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 425/500,0/9500
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = 425/500,0/50
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
stutter = 425/500,0/50
|
||||
|
||||
[no]
|
||||
description = Norway
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 425/200,0/600,425/200,0/10000
|
||||
dialrecall = 470/400,425/400
|
||||
record = 1400/400,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
|
||||
stutter = 470/400,425/400
|
||||
|
||||
[nz]
|
||||
description = New Zealand
|
||||
; Reference = http://www.telepermit.co.nz/TNA102.pdf
|
||||
ringcadence = 400,200,400,2000
|
||||
dial = 400
|
||||
busy = 400/500,0/500
|
||||
ring = 400+450/400,0/200,400+450/400,0/2000
|
||||
congestion = 400/250,0/250
|
||||
callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
|
||||
dialrecall = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
|
||||
record = 1400/425,0/15000
|
||||
info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
|
||||
stutter = !400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400
|
||||
unobtainable = 400/75,0/100,400/75,0/100,400/75,0/100,400/75,0/400
|
||||
|
||||
[ph]
|
||||
|
||||
; reference http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
|
||||
description = Philippines
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 480+620/500,0/500
|
||||
ring = 425+480/1000,0/4000
|
||||
congestion = 480+620/250,0/250
|
||||
callwaiting = 440/300,0/10000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
; INFO - not specified
|
||||
info = !950/330,!1400/330,!1800/330,0
|
||||
; STUTTER - not specified
|
||||
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
|
||||
|
||||
|
||||
[pl]
|
||||
description = Poland
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/500,0/500
|
||||
callwaiting = 425/150,0/150,425/150,0/4000
|
||||
; DIALRECALL - not specified
|
||||
dialrecall = 425/500,0/50
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
; 950/1400/1800 3x0.33 on 1.0 off repeated 3 times
|
||||
info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
|
||||
; STUTTER - not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[pt]
|
||||
description = Portugal
|
||||
ringcadence = 1000,5000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/5000
|
||||
congestion = 425/200,0/200
|
||||
callwaiting = 440/300,0/10000
|
||||
dialrecall = 425/1000,0/200
|
||||
record = 1400/500,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[ru]
|
||||
; References:
|
||||
; http://www.minsvyaz.ru/site.shtml?id=1806
|
||||
; http://www.aboutphone.info/lib/gost/45-223-2001.html
|
||||
description = Russian Federation / ex Soviet Union
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/350,0/350
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/175,0/175
|
||||
callwaiting = 425/200,0/5000
|
||||
record = 1400/400,0/15000
|
||||
info = 950/330,1400/330,1800/330,0/1000
|
||||
dialrecall = 425/400,0/40
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
[se]
|
||||
description = Sweden
|
||||
ringcadence = 1000,5000
|
||||
dial = 425
|
||||
busy = 425/250,0/250
|
||||
ring = 425/1000,0/5000
|
||||
congestion = 425/250,0/750
|
||||
callwaiting = 425/200,0/500,425/200,0/9100
|
||||
dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
record = 1400/500,0/15000
|
||||
info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
; stutter = 425/320,0/20 ; Real swedish standard, not used for now
|
||||
|
||||
[sg]
|
||||
description = Singapore
|
||||
; Singapore
|
||||
; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf
|
||||
; Frequency specs are: 425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
|
||||
ringcadence = 400,200,400,2000
|
||||
dial = 425
|
||||
ring = 425*24/400,0/200,425*24/400,0/2000 ; modulation should be 100%, not 90%
|
||||
busy = 425/750,0/750
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 425*24/300,0/200,425*24/300,0/3200
|
||||
stutter = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
|
||||
info = 950/330,1400/330,1800/330,0/1000 ; not currently in use acc. to reference
|
||||
dialrecall = 425*24/500,0/500,425/500,0/2500 ; unspecified in IDA reference, use repeating Holding Tone A,B
|
||||
record = 1400/500,0/15000 ; unspecified in IDA reference, use 0.5s tone every 15s
|
||||
; additionally defined in reference
|
||||
nutone = 425/2500,0/500
|
||||
intrusion = 425/250,0/2000
|
||||
warning = 425/624,0/4376 ; end of period tone, warning
|
||||
acceptance = 425/125,0/125
|
||||
holdinga = !425*24/500,!0/500 ; followed by holdingb
|
||||
holdingb = !425/500,!0/2500
|
||||
|
||||
[th]
|
||||
description = Thailand
|
||||
ringcadence = 1000,4000
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
dial = 400*50
|
||||
busy = 400/500,0/500
|
||||
ring = 420/1000,0/5000
|
||||
congestion = 400/300,0/300
|
||||
callwaiting = 1000/400,10000/400,1000/400
|
||||
; DIALRECALL - not specified - use special dial tone instead.
|
||||
dialrecall = 400*50/400,0/100,400*50/400,0/100
|
||||
; RECORDTONE - not specified
|
||||
record = 1400/500,0/15000
|
||||
; INFO - specified as an announcement - use special information tones instead
|
||||
info = 950/330,1400/330,1800/330
|
||||
; STUTTER - not specified
|
||||
stutter = !400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,!400/200,!0/200,!400/600,!0/200,400
|
||||
|
||||
[uk]
|
||||
description = United Kingdom
|
||||
ringcadence = 400,200,400,2000
|
||||
; These are the official tones taken from BT SIN350. The actual tones
|
||||
; used by BT include some volume differences so sound slightly different
|
||||
; from Asterisk-generated ones.
|
||||
dial = 350+440
|
||||
; Special dial is the intermittent dial tone heard when, for example,
|
||||
; you have a divert active on the line
|
||||
specialdial = 350+440/750,440/750
|
||||
; Busy is also called "Engaged"
|
||||
busy = 400/375,0/375
|
||||
; "Congestion" is the Beep-bip engaged tone
|
||||
congestion = 400/400,0/350,400/225,0/525
|
||||
; "Special Congestion" is not used by BT very often if at all
|
||||
specialcongestion = 400/200,1004/300
|
||||
unobtainable = 400
|
||||
ring = 400+450/400,0/200,400+450/400,0/2000
|
||||
callwaiting = 400/100,0/4000
|
||||
; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
|
||||
specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
|
||||
; "Pips" used by BT on payphones. (Sounds wrong, but this is what BT claim it
|
||||
; is and I've not used a payphone for years)
|
||||
creditexpired = 400/125,0/125
|
||||
; These two are used to confirm/reject service requests on exchanges that
|
||||
; don't do voice announcements.
|
||||
confirm = 1400
|
||||
switching = 400/200,0/400,400/2000,0/400
|
||||
; This is the three rising tones Doo-dah-dee "Special Information Tone",
|
||||
; usually followed by the BT woman saying an appropriate message.
|
||||
info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
|
||||
; Not listed in SIN350
|
||||
record = 1400/500,0/60000
|
||||
stutter = 350+440/750,440/750
|
||||
|
||||
[us]
|
||||
description = United States / North America
|
||||
ringcadence = 2000,4000
|
||||
dial = 350+440
|
||||
busy = 480+620/500,0/500
|
||||
ring = 440+480/2000,0/4000
|
||||
congestion = 480+620/250,0/250
|
||||
callwaiting = 440/300,0/10000
|
||||
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,0
|
||||
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
|
||||
[us-old]
|
||||
description = United States Circa 1950/ North America
|
||||
ringcadence = 2000,4000
|
||||
dial = 600*120
|
||||
busy = 500*100/500,0/500
|
||||
ring = 420*40/2000,0/4000
|
||||
congestion = 500*100/250,0/250
|
||||
callwaiting = 440/300,0/10000
|
||||
dialrecall = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,0
|
||||
stutter = !600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,!600*120/100,!0/100,600*120
|
||||
|
||||
[tw]
|
||||
description = Taiwan
|
||||
; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
|
||||
; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
|
||||
; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
|
||||
; http://www.pbx-manufacturer.com/ky120dx.htm
|
||||
; http://www.nettwerked.net/tones.txt
|
||||
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
|
||||
;
|
||||
; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
|
||||
; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
|
||||
; ringing tone 440+480Hz 1 sec. on ,2 sec. off
|
||||
;
|
||||
ringcadence = 1000,4000
|
||||
dial = 350+440
|
||||
busy = 480+620/500,0/500
|
||||
ring = 440+480/1000,0/2000
|
||||
congestion = 480+620/250,0/250
|
||||
callwaiting = 350+440/250,0/250,350+440/250,0/3250
|
||||
dialrecall = 300/1500,0/500
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1400/330,!1800/330,0
|
||||
stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
|
||||
|
||||
[ve]
|
||||
; Tone definition source for ve found on
|
||||
; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
|
||||
description = Venezuela / South America
|
||||
ringcadence = 1000,4000
|
||||
dial = 425
|
||||
busy = 425/500,0/500
|
||||
ring = 425/1000,0/4000
|
||||
congestion = 425/250,0/250
|
||||
callwaiting = 400+450/300,0/6000
|
||||
dialrecall = 425
|
||||
record = 1400/500,0/15000
|
||||
info = !950/330,!1440/330,!1800/330,0/1000
|
||||
; STUTTER - not specified
|
||||
stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
|
||||
|
||||
|
||||
[za]
|
||||
description = South Africa
|
||||
; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
|
||||
; (definitions for other countries can also be found there)
|
||||
; Note, though, that South Africa uses two switch types in their network --
|
||||
; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
|
||||
; The former use 383+417 in dial, ringback etc. The latter use 400*33
|
||||
; I've provided both, uncomment the ones you prefer
|
||||
ringcadence = 400,200,400,2000
|
||||
; dial/ring/callwaiting for the Siemens switches:
|
||||
dial = 400*33
|
||||
ring = 400*33/400,0/200,400*33/400,0/2000
|
||||
callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
|
||||
; dial/ring/callwaiting for the Alcatel switches:
|
||||
; dial = 383+417
|
||||
; ring = 383+417/400,0/200,383+417/400,0/2000
|
||||
; callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250
|
||||
congestion = 400/250,0/250
|
||||
busy = 400/500,0/500
|
||||
dialrecall = 350+440
|
||||
; XXX Not sure about the RECORDTONE
|
||||
record = 1400/500,0/10000
|
||||
info = 950/330,1400/330,1800/330,0/330
|
||||
stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33
|
||||
29
asterisk/keys/cert.pem
Executable file
29
asterisk/keys/cert.pem
Executable file
@ -0,0 +1,29 @@
|
||||
-----BEGIN CERTIFICATE-----
|
||||
MIIE8DCCA9igAwIBAgISBKY/Qa5A38fjFTyjvKUk+dzAMA0GCSqGSIb3DQEBCwUA
|
||||
MDIxCzAJBgNVBAYTAlVTMRYwFAYDVQQKEw1MZXQncyBFbmNyeXB0MQswCQYDVQQD
|
||||
EwJSMzAeFw0yMzExMTAxMTAwMTlaFw0yNDAyMDgxMTAwMThaMBwxGjAYBgNVBAMT
|
||||
EWRlbW8uZm94LmRpcGFsLnJ1MIIBIjANBgkqhkiG9w0BAQEFAAOCAQ8AMIIBCgKC
|
||||
AQEAzCTncwR6I7EzKttTM7YmMfEOg6VtOcNW3TpBWvFBBTVS3JZE2kvxnCSzvk9U
|
||||
xLAGhJfJJ1TFjVn+2lYEz/VhFzFq17V4IcFCK19L8gMeXlam5iKPpoRkL2C5ArGE
|
||||
Y42jomvx5LIlnF3EB0WOEg+jeLRTGcnzYfxnARbeU1+8Yu+ezryq/ptqd+Ao+f5z
|
||||
YGgWKbaa9bDeJEmbB85Kn/92BoVz/MB6xsAH8mF9u7tRDcjHekcFC7Vl1wh4F/XE
|
||||
9PyEsZ0CPqG992ua1sQBGeh0JmLXp7zUpdESBuoGtk2pW0KaH8V0G2+LIP9VW+D8
|
||||
a1mnmaurtoePqU/F0a3D1rXifwIDAQABo4ICFDCCAhAwDgYDVR0PAQH/BAQDAgWg
|
||||
MB0GA1UdJQQWMBQGCCsGAQUFBwMBBggrBgEFBQcDAjAMBgNVHRMBAf8EAjAAMB0G
|
||||
A1UdDgQWBBSeY1Qw9v2GS+prNIxHdFccNsHA7zAfBgNVHSMEGDAWgBQULrMXt1hW
|
||||
y65QCUDmH6+dixTCxjBVBggrBgEFBQcBAQRJMEcwIQYIKwYBBQUHMAGGFWh0dHA6
|
||||
Ly9yMy5vLmxlbmNyLm9yZzAiBggrBgEFBQcwAoYWaHR0cDovL3IzLmkubGVuY3Iu
|
||||
b3JnLzAcBgNVHREEFTATghFkZW1vLmZveC5kaXBhbC5ydTATBgNVHSAEDDAKMAgG
|
||||
BmeBDAECATCCAQUGCisGAQQB1nkCBAIEgfYEgfMA8QB3ADtTd3U+LbmAToswWwb+
|
||||
QDtn2E/D9Me9AA0tcm/h+tQXAAABi7kavYsAAAQDAEgwRgIhANuhpID17rxQMV+V
|
||||
bJDxIoq3jpssVORYxsrKL8XRGPCdAiEA8jAXp8WUdCENXe+GD2UbsLnA6IY3J7bE
|
||||
GugHwb5jXxcAdgDuzdBk1dsazsVct520zROiModGfLzs3sNRSFlGcR+1mwAAAYu5
|
||||
Gr2JAAAEAwBHMEUCIQCXctcG8wSlO8uceV0Go4EHSED4Kw0lGDpkxSiVmTFP5gIg
|
||||
GhPkleoM8l2sv/0kqBAqF32jRGlcgKg5ozfDKA6/M44wDQYJKoZIhvcNAQELBQAD
|
||||
ggEBAFPr0U27a2KTGVK06Wc/gOEYHSjE/NAVg64L/ZlspREJlIwGfCOedMjBXLPw
|
||||
d7vsqt4xLFrY251lRQqar5SIQtPuf9DLzf/B/l+Q9JbMlDM8QZQBxPdGRn6x4hF7
|
||||
FNiCKh6kLvzH8z3Vw0d435t56tJ70izAY/EHRw2+sxpYceFaR5ullc0NB9aA0n5F
|
||||
k7M00+4ta+YyCAMcquU/+5de3bviURmxDh7Sm4BmDRm+KdInjzbfcPLzfppgCzws
|
||||
v3dsWkAbnJCeKGOPjiRQjyBStczob6y7IFzKB9nKuqsL52Ii4m69wnL5/Pd+jKfr
|
||||
+8Cvy0Xv47P6v8VPzCbuD8KBO6Q=
|
||||
-----END CERTIFICATE-----
|
||||
30
asterisk/keys/cert1.pem
Executable file
30
asterisk/keys/cert1.pem
Executable file
@ -0,0 +1,30 @@
|
||||
-----BEGIN CERTIFICATE-----
|
||||
MIIFKTCCBBGgAwIBAgISBAv2+MFeWn74jIUP6nIql592MA0GCSqGSIb3DQEBCwUA
|
||||
MDIxCzAJBgNVBAYTAlVTMRYwFAYDVQQKEw1MZXQncyBFbmNyeXB0MQswCQYDVQQD
|
||||
EwJSMzAeFw0yMzAyMDIxNTU4MTFaFw0yMzA1MDMxNTU4MTBaMBwxGjAYBgNVBAMT
|
||||
EWRlbW8uZm94LmRpcGFsLnJ1MIIBIjANBgkqhkiG9w0BAQEFAAOCAQ8AMIIBCgKC
|
||||
AQEA1w6UsOZeTi4xSSHLd9leTZjsSFa+/n6dU/WP41Kp+sL3d8Nw0D5obtj6kLV0
|
||||
AOEnkJhmrh5bMWyeQZ2YkjjCVJVs38pMwHTloVBO+a8MZT8ZqA0Jwga0AaX1LVzL
|
||||
HJDbSb9COBLXX8JFQUn7K0ynourqA7JpUGTR8qRdd1dPeMgkKZfZcJ0afrPtAJ/e
|
||||
O6wqIL+bjq3rsr1f1js1/A7JPImS3O2iSPRv5bUliY5xEO0VPSFKdsEC8t+1pft4
|
||||
oIoVExmCEan5FhWf7cxCNwQ5FZqj0mGZvcxaU5U9yzpFkno4HyxBRS0nTsT4bPd7
|
||||
zfswQBkuzz+TvNVgGpbJH2zrqwIDAQABo4ICTTCCAkkwDgYDVR0PAQH/BAQDAgWg
|
||||
MB0GA1UdJQQWMBQGCCsGAQUFBwMBBggrBgEFBQcDAjAMBgNVHRMBAf8EAjAAMB0G
|
||||
A1UdDgQWBBRhiePmMYL9oBYWLVOkw/POrWEeaTAfBgNVHSMEGDAWgBQULrMXt1hW
|
||||
y65QCUDmH6+dixTCxjBVBggrBgEFBQcBAQRJMEcwIQYIKwYBBQUHMAGGFWh0dHA6
|
||||
Ly9yMy5vLmxlbmNyLm9yZzAiBggrBgEFBQcwAoYWaHR0cDovL3IzLmkubGVuY3Iu
|
||||
b3JnLzAcBgNVHREEFTATghFkZW1vLmZveC5kaXBhbC5ydTBMBgNVHSAERTBDMAgG
|
||||
BmeBDAECATA3BgsrBgEEAYLfEwEBATAoMCYGCCsGAQUFBwIBFhpodHRwOi8vY3Bz
|
||||
LmxldHNlbmNyeXB0Lm9yZzCCAQUGCisGAQQB1nkCBAIEgfYEgfMA8QB2ALc++yTf
|
||||
nE26dfI5xbpY9Gxd/ELPep81xJ4dCYEl7bSZAAABhhMQeZIAAAQDAEcwRQIgEm67
|
||||
Z/Hxjdz3yZiaXm4jL2FKgpkcazcTXdMPqgeqRBoCIQC77GnauWga1gZ3tI4q2bds
|
||||
vH3NxAJz+8gG+f1mvMzUtgB3AHoyjFTYty22IOo44FIe6YQWcDIThU070ivBOlej
|
||||
UutSAAABhhMQeaUAAAQDAEgwRgIhAMzds9QlNVv5aLZS3f5TD4aGJK5BL0FJN8VZ
|
||||
565dgDtyAiEAlo0H35gMbfYsYENN2Cx2hnwv8VSD5VdYxYKB7Q8f3AMwDQYJKoZI
|
||||
hvcNAQELBQADggEBAI4JLgCRfnN34+qeyrTi3jpAy8kPc9sHuWCZI8PhNIrFNFDX
|
||||
zpJbn1RRq2eQEdvyuWSDn02XOExrHXcwB2fPIAVYpNVWJSO+9zDjAFViZcADiJkj
|
||||
ej4TXN71cuVrJI5nQHC/cmwedRxdkNIsKKYqQBAzIqawyWY/M5mp18AfMtPVZebz
|
||||
2bWHsRBJXLREhOMfDAPijGwhmjXFaOklpRM1oNS83oqyfQfvRBc1AgGBqxEA9D3V
|
||||
w20Z9tuqBQhnc4CofGRok3rpzaVVWHCy5KdfTht+hGf+er9OSJICe8G6yaCPwwNK
|
||||
dJZEJ4tF030qRmm1KNUelMfs/rgGRVrzAVWRrNI=
|
||||
-----END CERTIFICATE-----
|
||||
28
asterisk/keys/privkey.pem
Executable file
28
asterisk/keys/privkey.pem
Executable file
@ -0,0 +1,28 @@
|
||||
-----BEGIN PRIVATE KEY-----
|
||||
MIIEvgIBADANBgkqhkiG9w0BAQEFAASCBKgwggSkAgEAAoIBAQDMJOdzBHojsTMq
|
||||
21MztiYx8Q6DpW05w1bdOkFa8UEFNVLclkTaS/GcJLO+T1TEsAaEl8knVMWNWf7a
|
||||
VgTP9WEXMWrXtXghwUIrX0vyAx5eVqbmIo+mhGQvYLkCsYRjjaOia/HksiWcXcQH
|
||||
RY4SD6N4tFMZyfNh/GcBFt5TX7xi757OvKr+m2p34Cj5/nNgaBYptpr1sN4kSZsH
|
||||
zkqf/3YGhXP8wHrGwAfyYX27u1ENyMd6RwULtWXXCHgX9cT0/ISxnQI+ob33a5rW
|
||||
xAEZ6HQmYtenvNSl0RIG6ga2TalbQpofxXQbb4sg/1Vb4PxrWaeZq6u2h4+pT8XR
|
||||
rcPWteJ/AgMBAAECggEADYOh6MzSkxtPclZbnzA/Pg4hslhYrzq9vN525iDTd5dM
|
||||
mXq/Kx5c68PIPGB/EZg/jjaII/jqLQ3Yn7aTEftCBqmIU6hZQA7NpRZlbiuDrqYY
|
||||
3WJLVTAte69w//slcvu06fMwa73ZXI0ehYa2vX5YiYySdcIXQRTW8s0DQcy0KW5q
|
||||
LqPr4TS1909895nS65s2+AzW01Hu1cBbqdBU1xFEZvA+hZKueoziezKNnUGcmQol
|
||||
q3z+2Wj1tvjsa/1p9orkDgVXjRRA7m+mdG3+Haddq+AzHmfirNgAXDZlHDdRFnu/
|
||||
ueuxjkkh9b76nz0QhK4I+KDbDT90stn9otn2w5Ow/QKBgQDuva09XwmAJ66kcNGX
|
||||
Z7jumeqhvJavSFCUIeS1tbtzB9B5aIDkTIcfO0oHgSiT5en+6oRDlrY44IR0pH9t
|
||||
zJ154+GiPY/6FJJDWH9mPpK5KRtB03pZsj9CTZq2gBb/7MlPVO7zpZ3QwIlLfp6J
|
||||
KUPmHN8B3sCQz1/ddK95CQWR3QKBgQDa5vP1oD5DaXNw8L58eTFKAYOFUyNwHBJv
|
||||
LpvcaFOuA2s5E+02Kj84i9Z6Kfqg6YPuMyPj3QlgXJbBGm5ZDGkn9UW9vaLrtFFH
|
||||
cFDntEYLhOJjqtY6WvBMZGJiBnO3OKtgWWcKLssxfRzE9hcfzvXD7TeaK6ebb8Xs
|
||||
22i0KEQ2CwKBgQC4bCmhWQL7zquoN8aQkP1Yw+luW0KD0Pb+F1iXzfUpnQW8Y6/9
|
||||
q+hNhf1f4rQ328lwtDTMCWZmdXmcqV4OsZuMhy7Lp4bLBo0eHQT9NKUfB9QYe4bu
|
||||
NzWKwGwdQsGBnaa8871K1csdcT4d5YgOKKRc5DyZMzvKIYfKiNLsSr3UuQKBgQCV
|
||||
R8z1I+tyI3azfaoZzrg7RN8BTBotuSclJT7rR0ZRtEkp6pnpR1RH5q1EMF6X38e1
|
||||
6LBAE5vyWBIC7Fh2Z81Kj6yY4bs6TITlWAYEjV77Ogr9dyKTNzMKlaa2PLorJ/1P
|
||||
PojTW+f7/ttgwoYwjf3yOFKj08OAb/AlP9eejJs/0wKBgFR0+cNX6ymqT1mkuOmJ
|
||||
8E6e3Lr1tg4mtsLW7wOOQsiZTsKda/JxfccCrHxLhss5iS9KFCwb+qoXO4L4/nbA
|
||||
xXymqu9HiNh/ezSefH7OFZItl32esdK4KliEkyWlRrsJxE/nNFJEAqIhHS2x3kpC
|
||||
GrBrNG/zvFNYiGwN+KS/rd8k
|
||||
-----END PRIVATE KEY-----
|
||||
28
asterisk/keys/privkey1.pem
Executable file
28
asterisk/keys/privkey1.pem
Executable file
@ -0,0 +1,28 @@
|
||||
-----BEGIN PRIVATE KEY-----
|
||||
MIIEvQIBADANBgkqhkiG9w0BAQEFAASCBKcwggSjAgEAAoIBAQDXDpSw5l5OLjFJ
|
||||
Ict32V5NmOxIVr7+fp1T9Y/jUqn6wvd3w3DQPmhu2PqQtXQA4SeQmGauHlsxbJ5B
|
||||
nZiSOMJUlWzfykzAdOWhUE75rwxlPxmoDQnCBrQBpfUtXMsckNtJv0I4EtdfwkVB
|
||||
SfsrTKei6uoDsmlQZNHypF13V094yCQpl9lwnRp+s+0An947rCogv5uOreuyvV/W
|
||||
OzX8Dsk8iZLc7aJI9G/ltSWJjnEQ7RU9IUp2wQLy37Wl+3igihUTGYIRqfkWFZ/t
|
||||
zEI3BDkVmqPSYZm9zFpTlT3LOkWSejgfLEFFLSdOxPhs93vN+zBAGS7PP5O81WAa
|
||||
lskfbOurAgMBAAECggEAAVn2RZL6W7u+6TU8/etP7uU7nYwXVipohbBg2kOuOBpa
|
||||
i2ufKAfZsydKPSI5cOSj/CzLqbjHckI6+lw/AF2P8bp1/TAHFxyE8Y8QzVE0Huds
|
||||
cK8S9R+tU5H96O47g9nwhCr0rmfb555PvhnKFMW+C0WLo+tOmHKRvoMZkpg/XRnD
|
||||
QHBj4Rp4zBEKsKQw2jXJhtExf73xKtwPPds9pHGqOQjEEwJC5eZ2UiZIJlZ44F6k
|
||||
kKtbkV4YGNOQVUOQIopcwx1T2/ykj7/n680yuATPWrxkShPNCUFACNhf440/F9qp
|
||||
JpcxUJ5t4U4dnup2oOWcgepUUKQfRcFHG2ri2tOAJQKBgQDeom4exv0JIKIbgBBZ
|
||||
BI8cMK1bsgCFoqE8esUsEFtzFOtz9goviBDeFG6KDSHbVmbRxYfBjiZrdUPEgn/F
|
||||
2KNd4bm7/HW8EGhA+rHVfH7Yz+BiY1GtGhw472kL5+34QNfmalC5O+YmxWu9+y8M
|
||||
80btBT7SuiSvu6Z7jgluVv3tPQKBgQD3SW6Hscx4onF9kxNoyU/7o5R80fB8QN5u
|
||||
n9lkxKiqXOBBOwMARtROaCjlEpIpqqlSS53yIbD95gsccpiQAoqh+AbLyuuVW5VV
|
||||
L4hVYxAtep5e+sPPY5Z26DfBoR2x2oqXYLoF7QysyMFk5sz2w1knCJqa4iwJ5Uh2
|
||||
66mtA64bBwKBgDF6fei/JpsxsnVUJ83w0nFcbBq126HXe+f6DPn3BKv12UIjSy43
|
||||
GjCQ7+QkeQLLF/zE3p/onKdDH3V73zO0KtKBiQq7V9sc9l6CmKUuju5WfLLBCfTj
|
||||
maQZqQnpWJzUpGbj997C7ipts708Jr/FDV09zjF9Bww7MwvAjF4GXVatAoGACfgJ
|
||||
3qqKRaq2zIsrUyQad1iG2elUuo2a6fbDus46xnNsDvUkD9TLJkYIsANppQhePxUR
|
||||
TvuQuQuXExSDlfqkQBdRntGVYbYJdR+vbVHLCypQTB/r2OZ44F5IMSQfoFXhFG0v
|
||||
0qFVy3lkYn7zcFc7ttkf8nJ/t/5oR3MEqhbl4+0CgYEAsOqKaOjn+02u0trT4QT+
|
||||
QmkiPXHpz5HlY4LkqlbRXov1hBks9Ot/RPbZEeQfryb1b9mXIzh+Hhys2QfpCQJR
|
||||
Lk5KhrKpJRqACp4d1nID1d1QvZ2PSwyMgtglnjr48D5fxXb0TsZRXz4UVvgA2Z30
|
||||
B+vUa3hGl+VHzvw7wIYCOYg=
|
||||
-----END PRIVATE KEY-----
|
||||
173
asterisk/logger.conf
Executable file
173
asterisk/logger.conf
Executable file
@ -0,0 +1,173 @@
|
||||
;
|
||||
; Logging Configuration
|
||||
;
|
||||
; In this file, you configure logging to files or to
|
||||
; the syslog system.
|
||||
;
|
||||
; "logger reload" at the CLI will reload configuration
|
||||
; of the logging system.
|
||||
|
||||
[general]
|
||||
;
|
||||
; Customize the display of debug message time stamps
|
||||
; this example is the ISO 8601 date format (yyyy-mm-dd HH:MM:SS)
|
||||
;
|
||||
; see strftime(3) Linux manual for format specifiers. Note that there is also
|
||||
; a fractional second parameter which may be used in this field. Use %1q
|
||||
; for tenths, %2q for hundredths, etc.
|
||||
;
|
||||
;dateformat=%F %T ; ISO 8601 date format
|
||||
;dateformat=%F %T.%3q ; with milliseconds
|
||||
;
|
||||
;
|
||||
; This makes Asterisk write callids to log messages
|
||||
; (defaults to yes)
|
||||
;use_callids = no
|
||||
;
|
||||
; This appends the hostname to the name of the log files.
|
||||
;appendhostname = yes
|
||||
;
|
||||
; This determines whether or not we log queue events to a file
|
||||
; (defaults to yes).
|
||||
;queue_log = no
|
||||
;
|
||||
; Determines whether the queue_log always goes to a file, even
|
||||
; when a realtime backend is present (defaults to no).
|
||||
;queue_log_to_file = yes
|
||||
;
|
||||
; Set the queue_log filename
|
||||
; (defaults to queue_log)
|
||||
;queue_log_name = queue_log
|
||||
;
|
||||
; When using realtime for the queue log, use GMT for the timestamp
|
||||
; instead of localtime. The default of this option is 'no'.
|
||||
;queue_log_realtime_use_gmt = yes
|
||||
;
|
||||
; Log rotation strategy:
|
||||
; none: Do not perform any logrotation at all. You should make
|
||||
; very sure to set up some external logrotate mechanism
|
||||
; as the asterisk logs can get very large, very quickly.
|
||||
; sequential: Rename archived logs in order, such that the newest
|
||||
; has the highest sequence number [default]. When
|
||||
; exec_after_rotate is set, ${filename} will specify
|
||||
; the new archived logfile.
|
||||
; rotate: Rotate all the old files, such that the oldest has the
|
||||
; highest sequence number [this is the expected behavior
|
||||
; for Unix administrators]. When exec_after_rotate is
|
||||
; set, ${filename} will specify the original root filename.
|
||||
; timestamp: Rename the logfiles using a timestamp instead of a
|
||||
; sequence number when "logger rotate" is executed.
|
||||
; When exec_after_rotate is set, ${filename} will
|
||||
; specify the new archived logfile.
|
||||
;rotatestrategy = rotate
|
||||
;
|
||||
; Run a system command after rotating the files. This is mainly
|
||||
; useful for rotatestrategy=rotate. The example allows the last
|
||||
; two archive files to remain uncompressed, but after that point,
|
||||
; they are compressed on disk.
|
||||
;
|
||||
; exec_after_rotate=gzip -9 ${filename}.2
|
||||
;
|
||||
;
|
||||
; For each file, specify what to log.
|
||||
;
|
||||
; For console logging, you set options at start of
|
||||
; Asterisk with -v for verbose and -d for debug
|
||||
; See 'asterisk -h' for more information.
|
||||
;
|
||||
; Directory for log files is configures in asterisk.conf
|
||||
; option astlogdir
|
||||
;
|
||||
; All log messages go to a queue serviced by a single thread
|
||||
; which does all the IO. This setting controls how big that
|
||||
; queue can get (and therefore how much memory is allocated)
|
||||
; before new messages are discarded.
|
||||
; The default is 1000
|
||||
;logger_queue_limit = 250
|
||||
;
|
||||
;
|
||||
[logfiles]
|
||||
;
|
||||
; Format is:
|
||||
;
|
||||
; logger_name => [formatter]levels
|
||||
;
|
||||
; The name of the logger dictates not only the name of the logging
|
||||
; channel, but also its type. Valid types are:
|
||||
; - 'console' - The root console of Asterisk
|
||||
; - 'syslog' - Linux syslog, with facilities specified afterwards with
|
||||
; a period delimiter, e.g., 'syslog.local0'
|
||||
; - 'filename' - The name of the log file to create. This is the default
|
||||
; for log channels.
|
||||
;
|
||||
; Filenames can either be relative to the standard Asterisk log directory
|
||||
; (see 'astlogdir' in asterisk.conf), or absolute paths that begin with
|
||||
; '/'.
|
||||
;
|
||||
; An optional formatter can be specified prior to the log levels sent
|
||||
; to the log channel. The formatter is defined immediately preceeding the
|
||||
; levels, and is enclosed in square brackets. Valid formatters are:
|
||||
; - [default] - The default formatter, this outputs log messages using a
|
||||
; human readable format.
|
||||
; - [plain] - The plain formatter, this outputs log messages using a
|
||||
; human readable format with the addition of function name
|
||||
; and line number. No color escape codes are ever printed
|
||||
; nor are verbose messages treated specially.
|
||||
; - [json] - Log the output in JSON. Note that JSON formatted log entries,
|
||||
; if specified for a logger type of 'console', will be formatted
|
||||
; per the 'default' formatter for log messages of type VERBOSE.
|
||||
; This is due to the remote consoles intepreting verbosity
|
||||
; outside of the logging subsystem.
|
||||
;
|
||||
; Log levels include the following, and are specified in a comma delineated
|
||||
; list:
|
||||
; debug
|
||||
; trace
|
||||
; notice
|
||||
; warning
|
||||
; error
|
||||
; verbose(<level>)
|
||||
; dtmf
|
||||
; fax
|
||||
; security
|
||||
;
|
||||
; Verbose takes an optional argument, in the form of an integer level.
|
||||
; Verbose messages with higher levels will not be logged to the file. If
|
||||
; the verbose level is not specified, it will log verbose messages following
|
||||
; the current level of the root console.
|
||||
;
|
||||
; Special level name "*" means all levels, even dynamic levels registered
|
||||
; by modules after the logger has been initialized (this means that loading
|
||||
; and unloading modules that create/remove dynamic logger levels will result
|
||||
; in these levels being included on filenames that have a level name of "*",
|
||||
; without any need to perform a 'logger reload' or similar operation).
|
||||
; Note that there is no value in specifying both "*" and specific level names
|
||||
; for a filename; the "*" level means all levels. The only exception is if
|
||||
; you need to specify a specific verbose level. e.g, "verbose(3),*".
|
||||
;
|
||||
; We highly recommend that you DO NOT turn on debug mode if you are simply
|
||||
; running a production system. Debug mode turns on a LOT of extra messages,
|
||||
; most of which you are unlikely to understand without an understanding of
|
||||
; the underlying code. Do NOT report debug messages as code issues, unless
|
||||
; you have a specific issue that you are attempting to debug. They are
|
||||
; messages for just that -- debugging -- and do not rise to the level of
|
||||
; something that merit your attention as an Asterisk administrator. Both
|
||||
; debug and trace messages are also very verbose and can and do fill up
|
||||
; logfiles quickly. This is another reason not to have debug or trace
|
||||
; modes on a production system unless you are in the process of debugging
|
||||
; a specific issue.
|
||||
;
|
||||
;debug => debug
|
||||
;trace => trace
|
||||
;security => security
|
||||
console => notice,warning,error
|
||||
;console => notice,warning,error,debug
|
||||
messages => notice,warning,error
|
||||
full => notice,warning,error,debug,verbose,dtmf,fax
|
||||
;
|
||||
;full-json => [json]debug,verbose,notice,warning,error,dtmf,fax
|
||||
;
|
||||
;syslog keyword : This special keyword logs to syslog facility
|
||||
;
|
||||
;syslog.local0 => notice,warning,error
|
||||
;
|
||||
164
asterisk/manager.conf
Executable file
164
asterisk/manager.conf
Executable file
@ -0,0 +1,164 @@
|
||||
;
|
||||
; AMI - The Asterisk Manager Interface
|
||||
;
|
||||
; Third party application call management support and PBX event supervision
|
||||
;
|
||||
; Use the "manager show commands" at the CLI to list available manager commands
|
||||
; and their authorization levels.
|
||||
;
|
||||
; "manager show command <command>" will show a help text.
|
||||
;
|
||||
; ---------------------------- SECURITY NOTE -------------------------------
|
||||
; Note that you should not enable the AMI on a public IP address. If needed,
|
||||
; block this TCP port with iptables (or another FW software) and reach it
|
||||
; with IPsec, SSH, or SSL vpn tunnel. You can also make the manager
|
||||
; interface available over http/https if Asterisk's http server is enabled in
|
||||
; http.conf and if both "enabled" and "webenabled" are set to yes in
|
||||
; this file. Both default to no. httptimeout provides the maximum
|
||||
; timeout in seconds before a web based session is discarded. The
|
||||
; default is 60 seconds.
|
||||
;
|
||||
[general]
|
||||
enabled = no
|
||||
webenabled = no
|
||||
|
||||
port = 5038
|
||||
bindaddr = 0.0.0.0
|
||||
|
||||
; Parameters that control AMI over TLS. ("enabled" must be set too).
|
||||
; You can open a connection to this socket with e.g.
|
||||
;
|
||||
; openssl s_client -connect my_host:5039
|
||||
;
|
||||
;tlsenable=no ; set to YES to enable it
|
||||
;tlsbindaddr=0.0.0.0:5039 ; address and port to bind to, default to bindaddr and port 5039
|
||||
;tlscertfile=/tmp/asterisk.pem ; path to the certificate.
|
||||
;tlsprivatekey=/tmp/private.pem ; path to the private key, if no private given,
|
||||
; if no tlsprivatekey is given, default is to search
|
||||
; tlscertfile for private key.
|
||||
;tlscipher=<cipher string> ; string specifying which SSL ciphers to use or not use
|
||||
;
|
||||
;allowmultiplelogin = yes ; IF set to no, rejects manager logins that are already in use.
|
||||
; ; The default is yes.
|
||||
;
|
||||
;displayconnects = yes
|
||||
;
|
||||
; Add a Unix epoch timestamp to events (not action responses)
|
||||
;
|
||||
;timestampevents = yes
|
||||
|
||||
;brokeneventsaction = yes ; Restore previous behavior that caused the events
|
||||
; action to not return a response in certain
|
||||
; circumstances. Defaults to 'no'.
|
||||
|
||||
;
|
||||
; Display certain channel variables every time a channel-oriented
|
||||
; event is emitted:
|
||||
;
|
||||
; Note that this does incur a performance penalty and should be avoided if possible.
|
||||
;
|
||||
;channelvars = var1,var2,var3
|
||||
|
||||
; debug = on ; enable some debugging info in AMI messages (default off).
|
||||
; Also accessible through the "manager debug" CLI command.
|
||||
|
||||
; authtimeout specifies the maximum number of seconds a client has to
|
||||
; authenticate. If the client does not authenticate beofre this timeout
|
||||
; expires, the client will be disconnected. (default: 30 seconds)
|
||||
|
||||
;authtimeout = 30
|
||||
|
||||
; authlimit specifies the maximum number of unauthenticated sessions that will
|
||||
; be allowed to connect at any given time.
|
||||
|
||||
;authlimit = 50
|
||||
|
||||
;httptimeout = 60
|
||||
; a) httptimeout sets the Max-Age of the http cookie
|
||||
; b) httptimeout is the amount of time the webserver waits
|
||||
; on a action=waitevent request (actually its httptimeout-10)
|
||||
; c) httptimeout is also the amount of time the webserver keeps
|
||||
; a http session alive after completing a successful action
|
||||
|
||||
|
||||
;[mark]
|
||||
;secret = mysecret
|
||||
;deny=0.0.0.0/0.0.0.0
|
||||
;permit=209.16.236.73/255.255.255.0
|
||||
;acl=named_acl_example ; use a named ACL from acl.conf
|
||||
;
|
||||
;
|
||||
;setvar=PBXACCOUNT=edvina
|
||||
; The setvar option defines channel variables that will be set when this account
|
||||
; originates a call. You can define multiple setvar= commands for one manager
|
||||
; user.
|
||||
;
|
||||
;eventfilter=Event: Newchannel
|
||||
;eventfilter=Channel: (PJ)?SIP/(james|jim|john)-
|
||||
;eventfilter=!Channel: DAHDI/
|
||||
; The eventfilter option is used to whitelist or blacklist events per user.
|
||||
; A filter consists of an (unanchored) regular expression that is run on the
|
||||
; entire event data. If the first character of the filter is an exclamation
|
||||
; mark (!), the filter is appended to the blacklist instead of the whitelist.
|
||||
; After first checking the read access below, the regular expression filters
|
||||
; are processed as follows:
|
||||
; - If no filters are configured all events are reported as normal.
|
||||
; - If there are white filters only: implied black all filter processed first,
|
||||
; then white filters.
|
||||
; - If there are black filters only: implied white all filter processed first,
|
||||
; then black filters.
|
||||
; - If there are both white and black filters: implied black all filter processed
|
||||
; first, then white filters, and lastly black filters.
|
||||
|
||||
;
|
||||
; If the device connected via this user accepts input slowly,
|
||||
; the timeout for writes to it can be increased to keep it
|
||||
; from being disconnected (value is in milliseconds)
|
||||
;
|
||||
; writetimeout = 100
|
||||
;
|
||||
;displayconnects = yes ; Display on CLI user login/logoff
|
||||
;
|
||||
; Authorization for various classes
|
||||
;
|
||||
; Read authorization permits you to receive asynchronous events, in general.
|
||||
; Write authorization permits you to send commands and get back responses. The
|
||||
; following classes exist:
|
||||
;
|
||||
; all - All event classes below (including any we may have missed).
|
||||
; system - General information about the system and ability to run system
|
||||
; management commands, such as Shutdown, Restart, and Reload. This
|
||||
; class also includes dialplan manipulation actions such as
|
||||
; DialplanExtensionAdd and DialplanExtensionRemove.
|
||||
; call - Information about channels and ability to set information in a
|
||||
; running channel.
|
||||
; log - Logging information. Read-only. (Defined but not yet used.)
|
||||
; verbose - Verbose information. Read-only. (Defined but not yet used.)
|
||||
; agent - Information about queues and agents and ability to add queue
|
||||
; members to a queue.
|
||||
; user - Permission to send and receive UserEvent.
|
||||
; config - Ability to read and write configuration files.
|
||||
; command - Permission to run CLI commands. Write-only.
|
||||
; dtmf - Receive DTMF events. Read-only.
|
||||
; reporting - Ability to get information about the system.
|
||||
; cdr - Output of cdr_manager, if loaded. Read-only.
|
||||
; dialplan - Receive NewExten and VarSet events. Read-only.
|
||||
; originate - Permission to originate new calls. Write-only.
|
||||
; agi - Output AGI commands executed. Input AGI command to execute.
|
||||
; cc - Call Completion events. Read-only.
|
||||
; aoc - Permission to send Advice Of Charge messages and receive Advice
|
||||
; - Of Charge events.
|
||||
; test - Ability to read TestEvent notifications sent to the Asterisk Test
|
||||
; Suite. Note that this is only enabled when the TEST_FRAMEWORK
|
||||
; compiler flag is defined.
|
||||
; security - Security Events. Read-only.
|
||||
; message - Permissions to send out of call messages. Write-only
|
||||
;
|
||||
;read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
|
||||
;write = system,call,agent,user,config,command,reporting,originate,message
|
||||
|
||||
|
||||
[admin]
|
||||
secret = thiswouldbeaninsecurepassword
|
||||
read = system,call,log,verbose,command,agent,config,read,write,originate
|
||||
write = system,call,log,verbose,command,agent,config,read,write,originate
|
||||
3
asterisk/manager.d/README.conf
Executable file
3
asterisk/manager.d/README.conf
Executable file
@ -0,0 +1,3 @@
|
||||
; Empty placeholder by the Debian packaging.
|
||||
; You can add manager users by dropping files with a .conf extension in
|
||||
; this directory.
|
||||
45
asterisk/meetme.conf
Executable file
45
asterisk/meetme.conf
Executable file
@ -0,0 +1,45 @@
|
||||
;
|
||||
; Configuration file for MeetMe simple conference rooms for Asterisk of course.
|
||||
;
|
||||
; This configuration file is read every time you call app meetme()
|
||||
|
||||
[general]
|
||||
;audiobuffers=32 ; The number of 20ms audio buffers to be used
|
||||
; when feeding audio frames from non-DAHDI channels
|
||||
; into the conference; larger numbers will allow
|
||||
; for the conference to 'de-jitter' audio that arrives
|
||||
; at different timing than the conference's timing
|
||||
; source, but can also allow for latency in hearing
|
||||
; the audio from the speaker. Minimum value is 2,
|
||||
; maximum value is 32.
|
||||
;
|
||||
; Conferences may be scheduled from realtime?
|
||||
;schedule=yes
|
||||
;
|
||||
; Update realtime when members login/out of the conference
|
||||
;logmembercount=no
|
||||
;
|
||||
; How much earlier than the start time should we allow participants to
|
||||
; join the conference (in seconds)?
|
||||
;fuzzystart=300
|
||||
;
|
||||
; If the participants join too early, how much time should we allow
|
||||
; to tell them that they've joined too early, rather than telling them
|
||||
; the conference simply doesn't exist (in seconds)?
|
||||
;earlyalert=3600
|
||||
;
|
||||
; How many seconds before the scheduled end of the conference should
|
||||
; the participants be warned?
|
||||
;endalert=120
|
||||
;
|
||||
[rooms]
|
||||
;
|
||||
; Usage is conf => confno[,pin][,adminpin]
|
||||
;
|
||||
; Note that once a participant has called the conference, a change to the pin
|
||||
; number done in this file will not take effect until there are no more users
|
||||
; in the conference and it goes away. When it is created again, it will have
|
||||
; the new pin number.
|
||||
;
|
||||
;conf => 1234
|
||||
;conf => 2345,9938
|
||||
142
asterisk/mgcp.conf
Executable file
142
asterisk/mgcp.conf
Executable file
@ -0,0 +1,142 @@
|
||||
;
|
||||
; MGCP Configuration for Asterisk
|
||||
;
|
||||
[general]
|
||||
;port = 2427
|
||||
;bindaddr = 0.0.0.0
|
||||
|
||||
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
|
||||
;tos=cs3 ; Sets TOS for signaling packets.
|
||||
;tos_audio=ef ; Sets TOS for RTP audio packets.
|
||||
;cos=3 ; Sets 802.1p priority for signaling packets.
|
||||
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
|
||||
|
||||
; --------------------- DIGIT TIMEOUTS ----------------------------
|
||||
firstdigittimeout = 30000 ; default 16000 = 16s
|
||||
gendigittimeout = 10000 ; default 8000 = 8s
|
||||
matchdigittimeout = 5000 ; defaults 3000 = 3s
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
||||
; MGCP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The MGCP channel can accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive MGCP side will only
|
||||
; be used if the sending side can create jitter and jbforce is
|
||||
; also set to yes.
|
||||
|
||||
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a MGCP
|
||||
; channel. Defaults to "no".
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a MGCP
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
;[dlinkgw]
|
||||
;host = 192.168.0.64
|
||||
;context = default
|
||||
;directmedia = no
|
||||
;line => aaln/2
|
||||
;line => aaln/1
|
||||
|
||||
;; The MGCP channel supports the following service codes:
|
||||
;; # - Transfer
|
||||
;; *67 - Calling Number Delivery Blocking
|
||||
;; *70 - Cancel Call Waiting
|
||||
;; *72 - Call Forwarding Activation
|
||||
;; *73 - Call Forwarding Deactivation
|
||||
;; *78 - Do Not Disturb Activation
|
||||
;; *79 - Do Not Disturb Deactivation
|
||||
;; *8 - Call pick-up
|
||||
;
|
||||
; known to work with Swissvoice IP10s
|
||||
;[192.168.1.20]
|
||||
;context=local
|
||||
;host=192.168.1.20
|
||||
;callerid = "John Doe" <123>
|
||||
;callgroup=0 ; in the range from 0 to 63
|
||||
;pickupgroup=0 ; in the range from 0 to 63
|
||||
;nat=no
|
||||
;threewaycalling=yes
|
||||
;transfer=yes ; transfer requires threewaycalling=yes. Use FLASH to transfer
|
||||
;callwaiting=yes ; this might be a cause of trouble for ip10s
|
||||
;cancallforward=yes
|
||||
;line => aaln/1
|
||||
;
|
||||
|
||||
;[dph100]
|
||||
;
|
||||
; Supporting the DPH100M requires defining DLINK_BUGGY_FIRMWARE in
|
||||
; chan_mgcp.c in addition to enabling the slowsequence mode due to
|
||||
; bugs in the D-Link firmware
|
||||
;
|
||||
;context=local
|
||||
;host=dynamic
|
||||
;dtmfmode=none ; DTMF Mode can be 'none', 'rfc2833', or 'inband' or
|
||||
; 'hybrid' which starts in none and moves to inband. Default is none.
|
||||
;slowsequence=yes ; The DPH100M does not follow MGCP standards for sequencing
|
||||
;line => aaln/1
|
||||
|
||||
; known to work with wave7optics FTTH LMGs
|
||||
;[192.168.1.20]
|
||||
;accountcode = 1000 ; record this in cdr as account identification for billing
|
||||
;amaflags = billing ; record this in cdr as flagged for 'billing',
|
||||
; 'documentation', or 'omit'
|
||||
;context = local
|
||||
;host = 192.168.1.20
|
||||
;wcardep = aaln/* ; enables wildcard endpoint and sets it to 'aaln/*'
|
||||
; another common format is '*'
|
||||
;callerid = "Duane Cox" <123> ; now lets setup line 1 using per endpoint configuration...
|
||||
;callwaiting = no
|
||||
;callreturn = yes
|
||||
;cancallforward = yes
|
||||
;directmedia = no
|
||||
;transfer = no
|
||||
;dtmfmode = inband
|
||||
;setvar=one=1 ; Set channel variables associated with this incoming line
|
||||
;setvar=two=2
|
||||
;line => aaln/1 ; now lets save this config to line1 aka aaln/1
|
||||
;clearvars=all ; Reset list of variables back to none
|
||||
;callerid = "Duane Cox" <456> ; now lets setup line 2
|
||||
;callwaiting = no
|
||||
;callreturn = yes
|
||||
;cancallforward = yes
|
||||
;directmedia = no
|
||||
;transfer = no
|
||||
;dtmfmode = inband
|
||||
;line => aaln/2 ; now lets save this config to line2 aka aaln/2
|
||||
|
||||
; PacketCable
|
||||
;[sbv5121e-mta.test.local]
|
||||
;host = 10.0.1.3
|
||||
;callwaiting = 1
|
||||
;canreinvite = 1
|
||||
;dtmfmode = rfc2833
|
||||
;amaflags = BILLING
|
||||
;ncs = yes ; Use NCS 1.0 signalling
|
||||
;pktcgatealloc = yes ; Allocate DQOS gate on CMTS
|
||||
;hangupongateremove = yes ; Hangup the channel if the CMTS close the gate
|
||||
;callerid = 3622622225
|
||||
;accountcode = test-3622622225
|
||||
;line = aaln/1
|
||||
;callerid = 3622622226
|
||||
;accountcode = test-3622622226
|
||||
;line = aaln/2
|
||||
218
asterisk/minivm.conf
Executable file
218
asterisk/minivm.conf
Executable file
@ -0,0 +1,218 @@
|
||||
;
|
||||
; Mini-Voicemail Configuration
|
||||
; for the MiniVM set of applications
|
||||
;
|
||||
; MiniVM consists of the following dialplan applications
|
||||
; MinivmGreet Play personal prompts for busy/unavailable/temporary messages or default prompts
|
||||
; MinivmRecord Record voice prompts to account directory or default directory
|
||||
; MinivmNotify Notify via e-mail or pager - with or without attachment
|
||||
; MinivmDelete Delete voice prompt (filename as argument or channel variable set by MinivmRecord)
|
||||
;
|
||||
; MiniVM works without accounts (just give e-mail address as argument) or with accounts in
|
||||
; this configuration file or realtime. The idea is to build voicemail as building blocks so that
|
||||
; a complete and adaptive voicemail system can be built in the dialplan
|
||||
;
|
||||
; ----------------------------- Variables to use in subject, from and message body ------------------
|
||||
; Change the from, body and/or subject, variables:
|
||||
; MVM_NAME, MVM_DUR, MVM_MSGNUM, VM_MAILBOX, MVM_CALLERID, MVM_CIDNUM,
|
||||
; MVM_CIDNAME, MVM_DATE
|
||||
;
|
||||
; In addition to these, you can set the MVM_COUNTER channel variable in the
|
||||
; dial plan and use that as a counter. It will also be used in the file name
|
||||
; of the media file attached to the message
|
||||
;
|
||||
; Note: The emailbody config row can only be up to 512 characters due to a
|
||||
; limitation in the Asterisk configuration subsystem.
|
||||
; To create longer mails, use the templatefile option when creating the template
|
||||
; ---------------------------------------------------------------------------------------------------
|
||||
|
||||
[general]
|
||||
; Default format for storing and sending voicemail
|
||||
; (only one format. Can also be set on a per-mailbox level)
|
||||
format=wav49
|
||||
;format=gsm
|
||||
;
|
||||
;Turn on logfile with the following syntax. One line per voicemail received
|
||||
;with minivmRecord()
|
||||
; Mailbox:domain:macrocontext:exten:priority:callerchan:callerid:origdate:origtime:duration:durationstatus:accountcode
|
||||
;logfile=/var/log/asterisk/minivm.log
|
||||
; Who the e-mail notification should appear to come from
|
||||
serveremail=asterisk
|
||||
;serveremail=asterisk@asterisk.example.com
|
||||
; Minimum length of a voicemail message in seconds for the message to be kept
|
||||
; The default is no minimum.
|
||||
;minmessage=3
|
||||
; How many seconds of silence before we end the recording
|
||||
maxsilence=10
|
||||
; Silence threshold (what we consider silence: the lower, the more sensitive)
|
||||
silencethreshold=128
|
||||
; How long greeting messages (busy/unavailable/temp/name) are allowed to be, in seconds
|
||||
;maxgreet=120
|
||||
; If you need to have an external program, i.e. /usr/bin/myapp called when a
|
||||
; voicemail is received by the server. The arguments are
|
||||
;
|
||||
; <app> <username@domain> <callerid-name> <callerid-number>
|
||||
;
|
||||
;externnotify=/usr/bin/myapp
|
||||
; The character set for voicemail messages can be specified here
|
||||
;charset=ISO-8859-1
|
||||
; Skip the "[PBX]:" string from the message title
|
||||
;pbxskip=yes
|
||||
; Change the From: string
|
||||
|
||||
; You can override the default program to send e-mail if you wish, too
|
||||
; This is used both for e-mail and pager messages
|
||||
;mailcmd=/usr/sbin/sendmail -t
|
||||
;
|
||||
; -------------Default e-mail message template (used if no templates are used) ------
|
||||
;fromstring=The Asterisk PBX
|
||||
;
|
||||
|
||||
;emailsubject=[PBX]: New message ${MVM_COUNER} in mailbox ${VM_MAILBOX}
|
||||
; The following definition is very close to the default, but the default shows
|
||||
; just the CIDNAME, if it is not null, otherwise just the CIDNUM, or "an unknown
|
||||
; caller", if they are both null.
|
||||
;emailbody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message (number ${MVM_COUNTER})\nin mailbox ${MVM_MAILBOX} from ${MVM_CALLERID}, on ${MVM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
|
||||
;
|
||||
; Set the date format on outgoing mails. Valid arguments can be found on the
|
||||
; strftime(3) man page
|
||||
;
|
||||
; Default
|
||||
emaildateformat=%A, %B %d, %Y at %r
|
||||
; 24h date format
|
||||
;emaildateformat=%A, %d %B %Y at %H:%M:%S
|
||||
;
|
||||
; -------------Default pager message template (used if no templates are used) ------
|
||||
; You can also change the Pager From: string, the pager body and/or subject.
|
||||
; The above defined variables also can be used here
|
||||
;pagerfromstring=The Asterisk PBX
|
||||
;pagersubject=New VM ${MVM_COUNTER}
|
||||
;pagerbody=New ${MVM_DUR} long msg in box ${MVM_MAILBOX}\nfrom ${MVM_CALLERID}, on ${MVM_DATE}
|
||||
;
|
||||
;
|
||||
; -------------Timezone definitions (used in voicemail accounts) -------------------
|
||||
;
|
||||
; Users may be located in different timezones, or may have different
|
||||
; message announcements for their introductory message when they enter
|
||||
; the voicemail system. Set the message and the timezone each user
|
||||
; hears here. Set the user into one of these zones with the tz= attribute
|
||||
; in the options field of the mailbox. Of course, language substitution
|
||||
; still applies here so you may have several directory trees that have
|
||||
; alternate language choices.
|
||||
;
|
||||
; Look in /usr/share/zoneinfo/ for names of timezones.
|
||||
; Look at the manual page for strftime for a quick tutorial on how the
|
||||
; variable substitution is done on the values below.
|
||||
;
|
||||
; Supported values:
|
||||
; 'filename' filename of a soundfile (single ticks around the filename
|
||||
; required)
|
||||
; ${VAR} variable substitution
|
||||
; A or a Day of week (Saturday, Sunday, ...)
|
||||
; B or b or h Month name (January, February, ...)
|
||||
; d or e numeric day of month (first, second, ..., thirty-first)
|
||||
; Y Year
|
||||
; I or l Hour, 12 hour clock
|
||||
; H Hour, 24 hour clock (single digit hours preceded by "oh")
|
||||
; k Hour, 24 hour clock (single digit hours NOT preceded by "oh")
|
||||
; M Minute, with 00 pronounced as "o'clock"
|
||||
; N Minute, with 00 pronounced as "hundred" (US military time)
|
||||
; P or p AM or PM
|
||||
; Q "today", "yesterday" or ABdY
|
||||
; (*note: not standard strftime value)
|
||||
; q "" (for today), "yesterday", weekday, or ABdY
|
||||
; (*note: not standard strftime value)
|
||||
; R 24 hour time, including minute
|
||||
;
|
||||
; The message here is not used in mini-voicemail, but stays for
|
||||
; backwards compatibility
|
||||
|
||||
[zonemessages]
|
||||
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
|
||||
central=America/Chicago|'vm-received' Q 'digits/at' IMp
|
||||
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
|
||||
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
|
||||
|
||||
; ---------------------- Message body templates---------------------
|
||||
; [template-name] ; "template-" is a verbatim marker
|
||||
; fromaddress = Your Friendly Asterisk Server
|
||||
; fromemail = asteriskvm@digium.com
|
||||
; subject = <string>
|
||||
; attachmedia = yes | no ; Add media file as attachment?
|
||||
; dateformat = <formatstring> ; See above
|
||||
; charset = <charset> ; Mime charset definition for e-mail messages
|
||||
; locale = <locale> ; Locale for LC_TIME - to get weekdays in local language
|
||||
; ; See your O/S documentation for proper settings for setlocale()
|
||||
; templatefile = <filename> ; File name (relative to Asterisk configuration directory,
|
||||
; or absolute
|
||||
; messagebody = Format ; Message body definition with variables
|
||||
;
|
||||
[template-sv_SE_email]
|
||||
messagebody=Hej ${MVM_NAME}:\n\n\tDu har fått ett röstbrevlåde-meddelande från ${MVM_CALLERID}.\nLängd: ${MVM_DUR}\nMailbox ${MVM_MAILBOX}\nDatum: ${MVM_DATE}. \nMeddelandet bifogas det här brevet. Om du inte kan läsa det, kontakta intern support. \nHälsningar\n\n\t\t\t\t--Asterisk\n
|
||||
subject = Du har fått röstmeddelande (se bilaga)
|
||||
fromemail = swedish-voicemail-service@stockholm.example.com
|
||||
fromaddress = Asterisk Röstbrevlåda
|
||||
charset=iso-8859-1
|
||||
attachmedia=yes
|
||||
dateformat=%A, %d %B %Y at %H:%M:%S
|
||||
locale=sv_SE
|
||||
|
||||
[template-en_US_email]
|
||||
messagebody=Dear ${MVM_NAME}:\n\n\tjust wanted to let you know you were just left a ${MVM_DUR} long message \nin mailbox ${MVM_MAILBOX} from ${MVM_CALLERID}, on ${MVM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n
|
||||
subject = New voicemail
|
||||
charset=ascii
|
||||
attachmedia=yes
|
||||
dateformat=%A, %B %d, %Y at %r
|
||||
|
||||
;[template-sv_SE_pager]
|
||||
;templatefile = templates/pager_sv_se.txt
|
||||
;subject = Du har fått voicemail
|
||||
;charset=iso-8859-1
|
||||
;attachmedia=no
|
||||
;locale=sv_SE
|
||||
|
||||
;[template-nb_NO_email]
|
||||
;templatefile = templates/email_nb_NO.txt
|
||||
;subject = Du har fått voicemail
|
||||
;charset=iso-8859-1
|
||||
;locale=nb_NO
|
||||
|
||||
;[template-en_US_email_southern]
|
||||
;templatefile = templates/email_en_US.txt
|
||||
;subject = Y'all got voicemail, honey!
|
||||
;charset=ascii
|
||||
|
||||
;[template-en_UK_email]
|
||||
;templatefile = templates/email_en_us.txt
|
||||
;subject = Dear old chap, you've got an electronic communique
|
||||
;charset=ascii
|
||||
|
||||
; ---------------------- Mailbox accounts --------------------------
|
||||
;Template for mailbox definition - all options
|
||||
;
|
||||
; [username@domain] ; Has to be unique within domain (MWM_USERNAME, MWM_DOMAIN)
|
||||
; etemplate = sv_SE ; Email template from [templates]
|
||||
; ptemplate = en_US ; Pager template from [templates]
|
||||
; email = userpart@domain ; Extra e-mail address (overrides mailbox name)
|
||||
; pager = pageremail@domain ; E-mail address for pager messages
|
||||
; fullname = Mark Spencer ; Full name (MWM_NAME)
|
||||
; options = ; E-mail options, se below
|
||||
; accountcode = ; Account code (read in dialplan function MINIVMACCOUNT)
|
||||
; pincode = ; Numeric pin code (read in dialplan function MINIVMACCOUNT)
|
||||
; timezone=se ; Time zone
|
||||
; serveremail = asterisk@digium.com ; Who to send email from (overrides template if set)
|
||||
; externnotify = <application> ; External application for this account
|
||||
; volgain = ; Volume gain setting (requires "sox")
|
||||
; setvar=SERVICENAME=Voop.com Networks ; Extra variables to use in template
|
||||
|
||||
; Remember that you can use Asterisk Configuration Templates (ACT)
|
||||
|
||||
; [template@example.com](!) ; Declare template
|
||||
; setvar=customerdomain=example.com
|
||||
; setvar=customerclass=gold
|
||||
; etemplate = sv_se_email
|
||||
; serveremail = voicemail@example.com
|
||||
|
||||
; [user2@example.com](template@example.com) ; Declare user2 account using template
|
||||
; fullname = Olle E. Johansson
|
||||
; ; User inherits everything from template
|
||||
537
asterisk/misdn.conf
Executable file
537
asterisk/misdn.conf
Executable file
@ -0,0 +1,537 @@
|
||||
;
|
||||
; chan_misdn sample config
|
||||
;
|
||||
|
||||
; general section:
|
||||
;
|
||||
; for debugging and general setup, things that are not bound to port groups
|
||||
;
|
||||
|
||||
[general]
|
||||
;
|
||||
; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
|
||||
;
|
||||
misdn_init=/etc/misdn-init.conf
|
||||
|
||||
; set debugging flag:
|
||||
; 0 - No Debug
|
||||
; 1 - mISDN Messages and * - Messages, and * - State changes
|
||||
; 2 - Messages + Message specific Informations (e.g. bearer capability)
|
||||
; 3 - very Verbose, the above + lots of Driver specific infos
|
||||
; 4 - even more Verbose than 3
|
||||
;
|
||||
; default value: 0
|
||||
;
|
||||
debug=0
|
||||
|
||||
|
||||
|
||||
; set debugging file and flags for mISDNuser (NT-Stack)
|
||||
;
|
||||
; flags can be or'ed with the following values:
|
||||
;
|
||||
; DBGM_NET 0x00000001
|
||||
; DBGM_MSG 0x00000002
|
||||
; DBGM_FSM 0x00000004
|
||||
; DBGM_TEI 0x00000010
|
||||
; DBGM_L2 0x00000020
|
||||
; DBGM_L3 0x00000040
|
||||
; DBGM_L3DATA 0x00000080
|
||||
; DBGM_BC 0x00000100
|
||||
; DBGM_TONE 0x00000200
|
||||
; DBGM_BCDATA 0x00000400
|
||||
; DBGM_MAN 0x00001000
|
||||
; DBGM_APPL 0x00002000
|
||||
; DBGM_ISDN 0x00004000
|
||||
; DBGM_SOCK 0x00010000
|
||||
; DBGM_CONN 0x00020000
|
||||
; DBGM_CDATA 0x00040000
|
||||
; DBGM_DDATA 0x00080000
|
||||
; DBGM_SOUND 0x00100000
|
||||
; DBGM_SDATA 0x00200000
|
||||
; DBGM_TOPLEVEL 0x40000000
|
||||
; DBGM_ALL 0xffffffff
|
||||
;
|
||||
|
||||
ntdebugflags=0
|
||||
ntdebugfile=/var/log/misdn-nt.log
|
||||
|
||||
|
||||
; some pbx systems do cut the L1 for some milliseconds, to avoid
|
||||
; dropping running calls, we can set this flag to yes and tell
|
||||
; mISDNuser not to drop the calls on L2_RELEASE
|
||||
ntkeepcalls=no
|
||||
|
||||
; the big trace
|
||||
;
|
||||
; default value: [not set]
|
||||
;
|
||||
;tracefile=/var/log/asterisk/misdn.log
|
||||
|
||||
|
||||
; set to yes if you want mISDN_dsp to bridge the calls in HW
|
||||
;
|
||||
; default value: yes
|
||||
;
|
||||
bridging=no
|
||||
|
||||
|
||||
; stops dialtone after getting first digit on nt Port
|
||||
;
|
||||
; default value: yes
|
||||
;
|
||||
stop_tone_after_first_digit=yes
|
||||
|
||||
; whether to append overlapdialed Digits to Extension or not
|
||||
;
|
||||
; default value: yes
|
||||
;
|
||||
append_digits2exten=yes
|
||||
|
||||
;;; CRYPTION STUFF
|
||||
|
||||
; Whether to look for dynamic crypting attempt
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
dynamic_crypt=no
|
||||
|
||||
; crypt_prefix, what is used for crypting Protocol
|
||||
;
|
||||
; default value: [not set]
|
||||
;
|
||||
crypt_prefix=**
|
||||
|
||||
; Keys for cryption, you reference them in the dialplan
|
||||
; later also in dynamic encr.
|
||||
;
|
||||
; default value: [not set]
|
||||
;
|
||||
crypt_keys=test,muh
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
|
||||
; SIP channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The SIP channel can accept jitter,
|
||||
; thus a jitterbuffer on the receive SIP side will be used only
|
||||
; if it is forced and enabled.
|
||||
|
||||
; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
|
||||
; channel. Defaults to "no".
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmaxsize) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
; users sections:
|
||||
;
|
||||
; name your sections as you wish but not "general" or "default" !
|
||||
; the sections are Groups, you can dial out in extensions.conf
|
||||
; with Dial(mISDN/g:extern/101) where extern is a section name,
|
||||
; chan_misdn tries every port in this section to find a
|
||||
; new free channel
|
||||
;
|
||||
; The default section is not a group section, it just contains config elements
|
||||
; which are inherited by group sections.
|
||||
;
|
||||
[default]
|
||||
|
||||
; define your default context here
|
||||
;
|
||||
; default value: default
|
||||
;
|
||||
context=misdn
|
||||
|
||||
; language
|
||||
;
|
||||
; default value: en
|
||||
;
|
||||
language=en
|
||||
|
||||
;
|
||||
; This option specifies a default music on hold class to
|
||||
; use when put on hold if the channel's moh class was not
|
||||
; explicitly set with Set(CHANNEL(musicclass)=whatever) and
|
||||
; the peer channel did not suggest a class to use.
|
||||
;
|
||||
musicclass=default
|
||||
|
||||
;
|
||||
; Either if we should produce DTMF Tones ourselves
|
||||
;
|
||||
senddtmf=yes
|
||||
|
||||
;
|
||||
; If we should generate Ringing for chan_sip and others
|
||||
;
|
||||
far_alerting=no
|
||||
|
||||
|
||||
;
|
||||
; Here you can list which bearer capabilities should be allowed:
|
||||
; all - allow any bearer capability
|
||||
; speech - allow speech
|
||||
; 3_1khz - allow 3.1KHz audio
|
||||
; digital_unrestricted - allow unrestricted digital
|
||||
; digital_restricted - allow restricted digital
|
||||
; video - allow video
|
||||
;
|
||||
; Example:
|
||||
; allowed_bearers=speech,3_1khz
|
||||
;
|
||||
allowed_bearers=all
|
||||
|
||||
; Incoming number prefixes for the indicated Type-Of-Number. These are
|
||||
; inserted before any number (caller, dialed, connected, redirecting,
|
||||
; redirection) received from the ISDN link if that number has the
|
||||
; corresponding Type-Of-Number.
|
||||
; See the dialplan options.
|
||||
;
|
||||
; default values:
|
||||
; unknownprefix=
|
||||
; internationalprefix=00
|
||||
; nationalprefix=0
|
||||
; netspecificprefix=
|
||||
; subscriberprefix=
|
||||
; abbreviatedprefix=
|
||||
;
|
||||
;unknownprefix=
|
||||
internationalprefix=00
|
||||
nationalprefix=0
|
||||
;netspecificprefix=
|
||||
;subscriberprefix=
|
||||
;abbreviatedprefix=
|
||||
|
||||
; set rx/tx gains between -8 and 8 to change the RX/TX Gain
|
||||
;
|
||||
; default values: rxgain: 0
|
||||
; txgain: 0
|
||||
;
|
||||
rxgain=0
|
||||
txgain=0
|
||||
|
||||
; some telcos especially in NL seem to need this set to yes, also in
|
||||
; switzerland this seems to be important
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
te_choose_channel=no
|
||||
|
||||
|
||||
|
||||
;
|
||||
; Monitors L1 of the port. If L1 is down it tries
|
||||
; to bring it up. The polling timeout is given in seconds.
|
||||
; Setting the value to 0 disables monitoring L1 of the port.
|
||||
;
|
||||
; default value: 0
|
||||
;
|
||||
; This option is only read at chan_misdn loading time.
|
||||
; You need to unload and load chan_misdn to change the
|
||||
; value. An asterisk restart will also do the trick.
|
||||
;
|
||||
l1watcher_timeout=0
|
||||
|
||||
;
|
||||
; This option defines, if chan_misdn should check the L1 on a PMP
|
||||
; before making a group call on it. The L1 may go down for PMP Ports
|
||||
; so we might need this.
|
||||
; But be aware! a broken or plugged off cable might be used for a group call
|
||||
; as well, since chan_misdn has no chance to distinguish if the L1 is down
|
||||
; because of a lost Link or because the Provider shut it down...
|
||||
;
|
||||
; default: no
|
||||
;
|
||||
pmp_l1_check=no
|
||||
|
||||
|
||||
;
|
||||
; in PMP this option defines which cause should be sent out to
|
||||
; the 3. caller. chan_misdn does not support callwaiting on TE
|
||||
; PMP side. This allows to modify the RELEASE_COMPLETE cause
|
||||
; at least.
|
||||
;
|
||||
reject_cause=16
|
||||
|
||||
|
||||
;
|
||||
; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
|
||||
; this requests additional Infos, so we can waitfordigits
|
||||
; without much issues. This works only for PTP Ports
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
need_more_infos=no
|
||||
|
||||
|
||||
;
|
||||
; set this to yes if you want to disconnect calls when a timeout occurs
|
||||
; for example during the overlapdial phase
|
||||
;
|
||||
nttimeout=no
|
||||
|
||||
; Set the method to use for channel selection:
|
||||
; standard - Use the first free channel starting from the lowest number.
|
||||
; standard_dec - Use the first free channel starting from the highest number.
|
||||
; round_robin - Use the round robin algorithm to select a channel. Use this
|
||||
; if you want to balance your load.
|
||||
;
|
||||
; default value: standard
|
||||
;
|
||||
method=standard
|
||||
|
||||
|
||||
; specify if chan_misdn should collect digits before going into the
|
||||
; dialplan, you can choose yes=4 Seconds, no, or specify the amount
|
||||
; of seconds you need;
|
||||
;
|
||||
overlapdial=yes
|
||||
|
||||
;
|
||||
; dialplan means Type Of Number in ISDN Terms
|
||||
; There are different types of the dialplan:
|
||||
;
|
||||
; dialplan -> for outgoing call's dialed number
|
||||
; localdialplan -> for outgoing call's callerid
|
||||
; (if -1 is set use the value from the asterisk channel)
|
||||
; cpndialplan -> for incoming call's connected party number sent to caller
|
||||
; (if -1 is set use the value from the asterisk channel)
|
||||
;
|
||||
; dialplan options:
|
||||
;
|
||||
; 0 - unknown
|
||||
; 1 - International
|
||||
; 2 - National
|
||||
; 3 - Network-Specific
|
||||
; 4 - Subscriber
|
||||
; 5 - Abbreviated
|
||||
;
|
||||
; default value: 0
|
||||
;
|
||||
dialplan=0
|
||||
localdialplan=0
|
||||
cpndialplan=0
|
||||
|
||||
|
||||
|
||||
;
|
||||
; turn this to no if you don't mind correct handling of Progress Indicators
|
||||
;
|
||||
early_bconnect=yes
|
||||
|
||||
|
||||
;
|
||||
; turn this on if you like to send Tone Indications to a Incoming
|
||||
; isdn channel on a TE Port. Rarely used, only if the Telco allows
|
||||
; you to send indications by yourself, normally the Telco sends the
|
||||
; indications to the remote party.
|
||||
;
|
||||
; default: no
|
||||
;
|
||||
incoming_early_audio=no
|
||||
|
||||
; uncomment the following to get into s extension at extension conf
|
||||
; there you can use DigitTimeout if you can't or don't want to use
|
||||
; isdn overlap dial.
|
||||
; note: This will jump into the s exten for every exten!
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
;always_immediate=no
|
||||
|
||||
;
|
||||
; set this to yes if you want to generate your own dialtone
|
||||
; with always_immediate=yes, else chan_misdn generates the dialtone
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
nodialtone=no
|
||||
|
||||
|
||||
; uncomment the following if you want callers which called exactly the
|
||||
; base number (so no extension is set) jump to the s extension.
|
||||
; if the user dials something more it jumps to the correct extension
|
||||
; instead
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
;immediate=no
|
||||
|
||||
; uncomment the following to have hold and retrieve support
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
;hold_allowed=yes
|
||||
|
||||
; Pickup and Callgroup
|
||||
;
|
||||
; default values: not set = 0
|
||||
; range: 0-63
|
||||
;
|
||||
;callgroup=1
|
||||
;pickupgroup=1
|
||||
|
||||
; Named pickup groups and named call groups
|
||||
;
|
||||
; give a name to groups and configure any number of groups
|
||||
;
|
||||
;namedcallgroup=engineering,sales,netgroup,protgroup
|
||||
;namedpickupgroup=sales
|
||||
|
||||
; Set the outgoing caller id to the value.
|
||||
;callerid="name" <number>
|
||||
|
||||
;
|
||||
; these are the exact isdn screening and presentation indicators
|
||||
; if -1 is given for either value the presentation indicators are used
|
||||
; from asterisks CALLERPRES function.
|
||||
; s=0, p=0 -> callerid presented
|
||||
; s=1, p=1 -> callerid restricted (the remote end does not see it!)
|
||||
;
|
||||
; default values s=-1, p=-1
|
||||
presentation=-1
|
||||
screen=-1
|
||||
|
||||
; Incoming calls will have a caller ID tag set to this value
|
||||
;
|
||||
;incoming_cid_tag = "asterisk"
|
||||
|
||||
; With this set, you can automatically append the MSN of a party
|
||||
; to the cid_tag. Incoming calls have the dialed number appended
|
||||
; to the tag, and outgoing calls have the caller number appended
|
||||
; to the tag. An '_' is used to separate the tag from the
|
||||
; MSN.
|
||||
; Default is no.
|
||||
;
|
||||
;append_msn_to_cid_tag = no
|
||||
|
||||
; Select what to do with outgoing COLP information on this port.
|
||||
;
|
||||
; 0 - Send out COLP information unaltered. (default)
|
||||
; 1 - Force COLP to restricted on all outgoing COLP information.
|
||||
; 2 - Do not send COLP information.
|
||||
outgoing_colp=0
|
||||
|
||||
; Put a display ie in the CONNECT message containing the following
|
||||
; information if it is available (nt port only):
|
||||
;
|
||||
; 0 - Do not put the connected line information in the display ie.
|
||||
; 1 - Put the available connected line name in the display ie.
|
||||
; 2 - Put the available connected line number in the display ie.
|
||||
; 3 - Put the available connected line name and number in the display ie.
|
||||
;
|
||||
display_connected=0
|
||||
|
||||
; Put a display ie in the SETUP message containing the following
|
||||
; information if it is available (nt port only):
|
||||
;
|
||||
; 0 - Do not put the caller information in the display ie.
|
||||
; 1 - Put the available caller name in the display ie.
|
||||
; 2 - Put the available caller number in the display ie.
|
||||
; 3 - Put the available caller name and number in the display ie.
|
||||
;
|
||||
display_setup=0
|
||||
|
||||
; This enables echo cancellation with the given number of taps.
|
||||
; Be aware: Move this setting only to outgoing portgroups!
|
||||
; A value of zero turns echo cancellation off.
|
||||
;
|
||||
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
|
||||
;
|
||||
; default value: no
|
||||
;
|
||||
;echocancel=no
|
||||
|
||||
;
|
||||
; chan_misdns jitterbuffer, default 4000
|
||||
;
|
||||
jitterbuffer=4000
|
||||
|
||||
;
|
||||
; change this threshold to enable dejitter functionality
|
||||
;
|
||||
jitterbuffer_upper_threshold=0
|
||||
|
||||
|
||||
;
|
||||
; change this to yes, if you want to bridge a mISDN data channel to
|
||||
; another channel type or to an application.
|
||||
;
|
||||
hdlc=no
|
||||
|
||||
|
||||
;
|
||||
; defines the maximum amount of incoming calls per port for
|
||||
; this group. Calls which exceed the maximum will be marked with
|
||||
; the channel variable MAX_OVERFLOW. It will contain the amount of
|
||||
; overflowed calls
|
||||
;
|
||||
max_incoming=-1
|
||||
|
||||
;
|
||||
; defines the maximum amount of outgoing calls per port for this group
|
||||
; exceeding calls will be rejected
|
||||
;
|
||||
max_outgoing=-1
|
||||
|
||||
;
|
||||
; Enable/disable the call-completion retention option support (ptp only).
|
||||
;
|
||||
; Note: To use the CCBS/CCNR supplementary service feature and other
|
||||
; supplementary services using FACILITY messages requires a
|
||||
; modified version of mISDN from:
|
||||
; http://svn.digium.com/svn/thirdparty/mISDN/trunk
|
||||
; http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
|
||||
;
|
||||
cc_request_retention=yes
|
||||
|
||||
[intern]
|
||||
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
|
||||
ports=1,2
|
||||
; context where to go to when incoming Call on one of the above ports
|
||||
context=Intern
|
||||
|
||||
[internPP]
|
||||
;
|
||||
; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
|
||||
; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
|
||||
; configs. For backwards compatibility you can still set ptp here.
|
||||
;
|
||||
ports=3
|
||||
|
||||
[first_extern]
|
||||
; again port defs
|
||||
ports=4
|
||||
; again a context for incoming calls
|
||||
context=Extern1
|
||||
; msns for te ports, listen on those numbers on the above ports, and
|
||||
; indicate the incoming calls to asterisk
|
||||
; here you can give a comma separated list or simply an '*' for
|
||||
; any msn.
|
||||
msns=*
|
||||
|
||||
; here an example with given msns
|
||||
[second_extern]
|
||||
ports=5
|
||||
context=Extern2
|
||||
callerid="Asterisk" <1234>
|
||||
msns=102,144,101,104
|
||||
55
asterisk/modules.conf
Executable file
55
asterisk/modules.conf
Executable file
@ -0,0 +1,55 @@
|
||||
;
|
||||
; Asterisk configuration file
|
||||
;
|
||||
; Module Loader configuration file
|
||||
;
|
||||
|
||||
[modules]
|
||||
autoload=yes
|
||||
;
|
||||
; Any modules that need to be loaded before the Asterisk core has been
|
||||
; initialized (just after the logger initialization) can be loaded
|
||||
; using 'preload'. 'preload' forces a module and the modules it
|
||||
; is known to depend upon to be loaded earlier than they normally get
|
||||
; loaded.
|
||||
;
|
||||
; NOTE: There is no good reason left to use 'preload' anymore. It was
|
||||
; historically required to preload realtime driver modules so you could
|
||||
; map Asterisk core configuration files to Realtime storage.
|
||||
; This is no longer needed.
|
||||
;
|
||||
;preload => your_special_module.so
|
||||
;
|
||||
; If you want Asterisk to fail if a module does not load, then use
|
||||
; the "require" keyword. Asterisk will exit with a status code of 2
|
||||
; if a required module does not load.
|
||||
;
|
||||
; require = chan_pjsip.so
|
||||
;
|
||||
; If you want you can combine with preload
|
||||
; preload-require = your_special_module.so
|
||||
;
|
||||
;load => res_musiconhold.so
|
||||
;
|
||||
; Load one of: chan_oss, alsa, or console (portaudio).
|
||||
; By default, load chan_oss only (automatically).
|
||||
;
|
||||
noload => chan_alsa.so
|
||||
;noload => chan_oss.so
|
||||
noload => chan_console.so
|
||||
|
||||
noload => res_hep.so
|
||||
noload => res_hep_pjsip.so
|
||||
noload => res_hep_rtcp.so
|
||||
|
||||
; Do not load chan_sip by default, it may conflict with res_pjsip.
|
||||
noload => chan_sip.so
|
||||
;noload => res_srtp.so
|
||||
;noload => res_pjsip.so
|
||||
|
||||
; The default voicemail module is app_voicemal. All voicemail modules
|
||||
; are mutually exclusive. Therefore it is better to make sure they
|
||||
; are not loaded at startup
|
||||
;
|
||||
noload => app_voicemail_odbc.so
|
||||
noload => app_voicemail_imap.so
|
||||
99
asterisk/motif.conf
Executable file
99
asterisk/motif.conf
Executable file
@ -0,0 +1,99 @@
|
||||
; Sample configuration file for chan_motif
|
||||
|
||||
; Transports
|
||||
;
|
||||
; There are three different transports and protocol derivatives supported by chan_motif. They are in order of preference:
|
||||
; Jingle using ICE-UDP, Google Jingle, and Google-V1.
|
||||
;
|
||||
; Jingle as defined in XEP-0166 supports the widest range of features. It is referred to as "ice-udp" in this file. This is
|
||||
; the specification that Jingle clients implement.
|
||||
;
|
||||
; Google Jingle follows the Jingle specification for signaling but uses a custom transport for media. It is supported
|
||||
; by the Google Talk Plug-in in Gmail and by some other Jingle clients. It is referred to as "google" in this file.
|
||||
;
|
||||
; Google-V1 is the original Google Talk signaling protocol which uses an initial preliminary version of Jingle.
|
||||
; It also uses the same custom transport as Google Jingle for media. It is supported by Google Voice, some other Jingle
|
||||
; clients, and the Windows Google Talk client. It is referred to as "google-v1" in this file.
|
||||
;
|
||||
; Incoming sessions will automatically switch to the correct transport once it has been determined.
|
||||
;
|
||||
; Outgoing sessions are capable of determining if the target is capable of Jingle or a Google transport if the target is
|
||||
; in the roster. Unfortunately it is not possible to differentiate between a Google Jingle or Google-V1 capable resource
|
||||
; until a session initiate attempt occurs. If a resource is determined to use a Google transport it will initially use
|
||||
; Google Jingle but will fall back to Google-V1 if required.
|
||||
;
|
||||
; If an outgoing session attempt fails due to failure to support the given transport chan_motif will fall back in preference
|
||||
; order listed at the beginning of this document until all transports have been exhausted.
|
||||
;
|
||||
|
||||
; Dialing and Resource Selection Strategy
|
||||
;
|
||||
; Placing a call through an endpoint can be accomplished using the following dial string:
|
||||
;
|
||||
; Motif/<endpoint name>/<target>
|
||||
;
|
||||
; When placing an outgoing call through an endpoint the requested target is searched for in the roster list. If present
|
||||
; the first Jingle or Google Jingle capable resource is specifically targetted. Since the capabilities of the resource are
|
||||
; known the outgoing session initation will disregard the configured transport and use the determined one.
|
||||
;
|
||||
; If the target is not found in the roster the target will be used as-is and a session will be initiated using the
|
||||
; transport specified in this configuration file. If no transport has been specified the endpoint defaults to ice-udp.
|
||||
;
|
||||
|
||||
; Video Support
|
||||
;
|
||||
; Support for video does not need to be explicitly enabled. Configuring any video codec on your endpoint will
|
||||
; automatically enable it.
|
||||
|
||||
; DTMF
|
||||
;
|
||||
; The only supported method for DTMF is RFC2833. This is always enabled on audio streams and negotiated if possible.
|
||||
|
||||
; Incoming Calls
|
||||
;
|
||||
; Incoming calls will first look for the extension matching the name of the endpoint in the configured context. If
|
||||
; no such extension exists the call will automatically fall back to the "s" extension.
|
||||
|
||||
; CallerID
|
||||
;
|
||||
; The incoming caller id number is populated with the username of the caller and the name is populated with the full
|
||||
; identity of the caller. If you would like to perform authentication or filtering of incoming calls it is recommended
|
||||
; that you use these fields to do so.
|
||||
;
|
||||
; Outgoing caller id can *not* be set.
|
||||
;
|
||||
; IMPORTANT: Multiple endpoints using the same connection is *NOT* supported. Doing so may result in broken calls.
|
||||
|
||||
; Default template for endpoints, to be included in their definition
|
||||
[default](!)
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
allow=h264
|
||||
context=incoming-motif ; Default context that incoming sessions will land in
|
||||
|
||||
;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer
|
||||
;maxpayloads = 30 ; Maximum number of payloads we will offer
|
||||
|
||||
; Sample configuration entry for Jingle
|
||||
;[jingle-endpoint](default)
|
||||
;transport=ice-udp ; Change the default protocol of outgoing sessions to Jingle ICE-UDP
|
||||
;allow=g722 ; Add G.722 as an allowed format since the other side may support it
|
||||
;connection=local-jabber-account ; Connection to accept traffic on and send traffic out
|
||||
;accountcode=jingle ; Account code for CDR purposes
|
||||
|
||||
; Sample configuration entry for Google Talk
|
||||
[gtalk-endpoint](default)
|
||||
;transport=google ; Since this is a Google Talk endpoint we want to offer Google Jingle for outgoing sessions
|
||||
;connection=gtalk-account
|
||||
|
||||
; Sample configuration entry for Google Voice
|
||||
;[gvoice](default)
|
||||
;transport=google-v1 ; Google Voice uses the original Google Talk protocol
|
||||
;connection=gvoice-account
|
||||
|
||||
; Additional options
|
||||
; callgroup
|
||||
; pickupgroup
|
||||
; language
|
||||
; musicclass
|
||||
; parkinglot
|
||||
145
asterisk/musiconhold.conf
Executable file
145
asterisk/musiconhold.conf
Executable file
@ -0,0 +1,145 @@
|
||||
;
|
||||
; Music on Hold -- Sample Configuration
|
||||
;
|
||||
[general]
|
||||
;cachertclasses=yes ; use 1 instance of moh class for all users who are using it,
|
||||
; decrease consumable cpu cycles and memory
|
||||
; disabled by default
|
||||
|
||||
;preferchannelclass=yes ; Prefer the musicclass as defined by CHANNEL(musicclass),
|
||||
; over a class set by an application (e.g. a musicclass set on a queue).
|
||||
; Defaults to yes.
|
||||
|
||||
; valid mode options:
|
||||
; files -- read files from a directory in any Asterisk supported
|
||||
; media format
|
||||
; playlist -- provide a fixed list of filenames or HTTP(S) URLs to play
|
||||
; quietmp3 -- default
|
||||
; mp3 -- loud
|
||||
; mp3nb -- unbuffered
|
||||
; quietmp3nb -- quiet unbuffered
|
||||
; custom -- run a custom application (See examples below)
|
||||
|
||||
; =========
|
||||
; File-based (native) music on hold
|
||||
; =========
|
||||
;
|
||||
; This plays files directly from the specified directory, no external
|
||||
; processes are required. Files are played in normal sorting order
|
||||
; (same as a sorted directory listing), and no volume or other
|
||||
; sound adjustments are available. If the file is available in
|
||||
; the same format as the channel's codec, then it will be played
|
||||
; without transcoding (same as Playback would do in the dialplan).
|
||||
; Files can be present in as many formats as you wish, and the
|
||||
; 'best' format will be chosen at playback time.
|
||||
;
|
||||
; The path specified can be either an absolute path (starts with '/'),
|
||||
; or a relative path; relative paths are interpreted as being relative
|
||||
; to the 'astdatalibdir' in asterisk.conf, which defaults to
|
||||
; /var/lib/asterisk.
|
||||
;
|
||||
; NOTE:
|
||||
; If you are not using "autoload" in modules.conf, then you
|
||||
; must ensure that the format modules for any formats you wish
|
||||
; to use are loaded _before_ res_musiconhold. If you do not do
|
||||
; this, res_musiconhold will skip the files it is not able to
|
||||
; understand when it loads.
|
||||
;
|
||||
; =========
|
||||
; Playlist (native) music on hold
|
||||
; =========
|
||||
;
|
||||
; This mode is similar to 'files' mode in that it plays through a list
|
||||
; of files, but instead of scanning a directory the files are
|
||||
; explicitly configured using one or more 'entry' options.
|
||||
;
|
||||
; Each entry must be one of:
|
||||
;
|
||||
; * An absolute path to the file to be played, without an extension.
|
||||
; * An HTTP(S) URL
|
||||
;
|
||||
; The entries are played in the order in which they appear in the
|
||||
; configuration. The 'sort' option is not used for this mode.
|
||||
;
|
||||
|
||||
[default]
|
||||
mode=files
|
||||
directory=moh
|
||||
;
|
||||
;[native-random]
|
||||
;mode=files
|
||||
;directory=moh
|
||||
;digit=# ; If this option is set for a class, then when callers are
|
||||
; ; listening to music on hold, they can press this digit, and
|
||||
; ; they will switch to listening to this music class.
|
||||
;announcement=queue-thankyou ;If this option is set for a class, then
|
||||
; ; when callers get put on hold, the specified sound will be
|
||||
; ; be played to them. Also, if using modes that Asterisk
|
||||
; ; controls the playlist for (files, mp3, etc), the same
|
||||
; ; sound will also be played between MOH songs.
|
||||
;sort=random ; Sort the files in random order. Other sort options are:
|
||||
; ; 'alpha' and 'randstart'. If 'alpha', the files are sorted
|
||||
; ; in alphabetical order. If 'randstart', the files are sorted
|
||||
; ; in alphabetical order as well, but the first file is chosen
|
||||
; ; at random. If unspecified, the sort order is undefined.
|
||||
|
||||
;[native-alphabetical]
|
||||
;mode=files
|
||||
;directory=moh
|
||||
;sort=alpha ; Sort the files in alphabetical order.
|
||||
|
||||
;[sales-queue-hold]
|
||||
;mode=playlist
|
||||
;entry=/var/lib/asterisk/sounds/en/yourcallisimportant
|
||||
;entry=http://example.local/sales-queue-hold-music.ulaw
|
||||
;entry=/var/lib/asterisk/moh/macroform-robot_dity
|
||||
|
||||
; =========
|
||||
; Other (non-native) playback methods
|
||||
; =========
|
||||
|
||||
;[manual]
|
||||
;mode=custom
|
||||
; Note that with mode=custom, a directory is not required, such as when reading
|
||||
; from a stream.
|
||||
;directory=/var/lib/asterisk/mohmp3
|
||||
;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
|
||||
|
||||
;[ulawstream]
|
||||
;mode=custom
|
||||
;application=/usr/bin/streamplayer 192.168.100.52 888
|
||||
;format=ulaw ; The 'format' option specifies the audio format that the
|
||||
; ; 'application' will provide to Asterisk. In this example,
|
||||
; ; streamplayer will output ulaw samples so we need to set the
|
||||
; ; format to ulaw so that Asterisk knows how to interpret the
|
||||
; ; incoming audio.
|
||||
|
||||
; mpg123 on Solaris does not always exit properly; madplay may be a better
|
||||
; choice
|
||||
;[solaris]
|
||||
;mode=custom
|
||||
;directory=/var/lib/asterisk/mohmp3
|
||||
;application=/site/sw/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
|
||||
|
||||
; By default, when res_musiconhold reloads or unloads, it sends a HUP signal
|
||||
; to custom applications (and all descendants), waits 100ms, then sends a
|
||||
; TERM signal, waits 100ms, then finally sends a KILL signal. An application
|
||||
; which is interacting with an external device and/or spawns children of its
|
||||
; own may not be able to exit cleanly in the default times, expecially if sent
|
||||
; a KILL signal, or if it's children are getting signals directly from
|
||||
; res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
|
||||
; class option can be used to set the number of milliseconds res_musiconhold
|
||||
; waits before escalating kill signals, with the default being the current
|
||||
; 100ms. To control to whom the signals are sent, the "kill_method"
|
||||
; class option can be set to "process_group" (the default, existing behavior),
|
||||
; which sends signals to the application and its descendants directly, or
|
||||
; "process" which sends signals only to the application itself.
|
||||
|
||||
;[sox_from_device]
|
||||
;mode=custom
|
||||
;directory=/var/lib/asterisk/mohmp3
|
||||
;application=/usr/bin/sox -q -t alsa -c 2 -r 48000 hw:1 -c 1 -r 8000 -t raw -s -
|
||||
; Wait 500ms before escalating kill signals
|
||||
;kill_escalation_delay=500
|
||||
; Send signals to just the child process instead of all descendants
|
||||
;kill_method=process
|
||||
39
asterisk/muted.conf
Executable file
39
asterisk/muted.conf
Executable file
@ -0,0 +1,39 @@
|
||||
#
|
||||
# Sample muted configuration file
|
||||
#
|
||||
# Copyright (C) 2004 Digium, Inc.
|
||||
#
|
||||
# First you have the host, username, and password
|
||||
# we use to connect to the asterisk system
|
||||
#
|
||||
# What is this? Well, haven't you ever wished you could automatically
|
||||
# turn down the volume on your stereo, CDPlayer, etc, when a call comes in,
|
||||
# and then return it to normal when the call ends? Well, this is a possible
|
||||
# mechanism to make this happen!
|
||||
# You have to fire up the new utils/muted, which runs as a daemon in the
|
||||
# background. This daemon connects to asterisk via a manager interface, and
|
||||
# also reads this config file from /etc/muted.conf. when the channels mentioned
|
||||
# are activated, it tweaks the sound levels on the sound card(s).
|
||||
# So, depending on the sound card, you may be able to run all your sound
|
||||
# generating devices thru your sound card, and use this mechanism to quiet
|
||||
# them down while you are on the phone. If anyone figures out how to make
|
||||
# this work with kids, please inform!!
|
||||
#
|
||||
host localhost
|
||||
user user
|
||||
pass pass
|
||||
#
|
||||
# List each channel we're supposed to watch
|
||||
#
|
||||
channel DAHDI/1
|
||||
channel DAHDI/2
|
||||
channel SIP/mark
|
||||
#
|
||||
# Mute level is the percentage of the current volume we should
|
||||
# lower the music to.
|
||||
#
|
||||
mutelevel 20
|
||||
#
|
||||
# Smooth fade makes the fadein/fadeout nicer sounding
|
||||
#
|
||||
smoothfade
|
||||
203
asterisk/ooh323.conf
Executable file
203
asterisk/ooh323.conf
Executable file
@ -0,0 +1,203 @@
|
||||
; ---------------------------------------------------------------------------------
|
||||
; --- ******* IMPORTANT NOTE ***********
|
||||
; ---
|
||||
; --- This module is currently unsupported. Use it at your own risk.
|
||||
; ---
|
||||
; ---------------------------------------------------------------------------------
|
||||
|
||||
; Objective System's H323 Configuration example for Asterisk
|
||||
; ooh323c driver configuration
|
||||
;
|
||||
; [general] section defines global parameters
|
||||
;
|
||||
; This is followed by profiles which can be of three types - user/peer/friend
|
||||
; Name of the user profile should match with the h323id of the user device.
|
||||
; For peer/friend profiles, host ip address must be provided as "dynamic" is
|
||||
; not supported as of now.
|
||||
;
|
||||
; Syntax for specifying a H323 device in extensions.conf is
|
||||
; For Registered peers/friends profiles:
|
||||
; OOH323/name where name is the name of the peer/friend profile.
|
||||
;
|
||||
; For unregistered H.323 phones:
|
||||
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
|
||||
; alias
|
||||
;
|
||||
; For dialing into another asterisk peer at a specific exten
|
||||
; OOH323/exten/peer OR OOH323/exten@ip
|
||||
;
|
||||
; Domain name resolution is not yet supported.
|
||||
;
|
||||
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
|
||||
; name and context is determined to route the call
|
||||
;
|
||||
; The channel driver will register all global aliases and aliases defined in
|
||||
; peer profiles with the gatekeeper, if one exists. So, that when someone
|
||||
; outside our pbx (non-user) calls an extension, gatekeeper will route that
|
||||
; call to our asterisk box, from where it will be routed as per dial plan.
|
||||
|
||||
|
||||
[general]
|
||||
;Define the asetrisk server h323 endpoint
|
||||
|
||||
;The port asterisk should listen for incoming H323 connections.
|
||||
;Default - 1720
|
||||
;port=1720
|
||||
|
||||
;The dotted IP address asterisk should listen on for incoming H323
|
||||
;connections
|
||||
;Default - tries to find out local ip address on it's own
|
||||
bindaddr=0.0.0.0
|
||||
|
||||
;This parameter indicates whether channel driver should register with
|
||||
;gatekeeper as a gateway or an endpoint.
|
||||
;Default - no
|
||||
;gateway=no
|
||||
|
||||
;Whether asterisk should use fast-start and tunneling for H323 connections.
|
||||
;Default - yes
|
||||
;faststart=no
|
||||
;h245tunneling=no
|
||||
|
||||
;Whether media wait for connect
|
||||
;Default - No
|
||||
;mediawaitforconnect=yes
|
||||
|
||||
;H323-ID to be used for asterisk server
|
||||
;Default - Asterisk PBX
|
||||
h323id=ObjSysAsterisk
|
||||
e164=100
|
||||
|
||||
;CallerID to use for calls
|
||||
;Default - Same as h323id
|
||||
callerid=asterisk
|
||||
|
||||
; Whether asterisk send back to caller own connected line id on incoming call as called number
|
||||
; Default - no
|
||||
aniasdni=no
|
||||
|
||||
;Whether this asterisk server will use gatekeeper.
|
||||
;Default - DISABLE
|
||||
;gatekeeper = DISCOVER
|
||||
;gatekeeper = a.b.c.d
|
||||
gatekeeper = DISABLE
|
||||
|
||||
;Location for H323 log file
|
||||
;Default - /var/log/asterisk/h323_log
|
||||
;logfile=/var/log/asterisk/h323_log
|
||||
|
||||
|
||||
;Following values apply to all users/peers/friends defined below, unless
|
||||
;overridden within their client definition
|
||||
|
||||
;Sets default context all clients will be placed in.
|
||||
;Default - default
|
||||
context=default
|
||||
|
||||
;Sets rtptimeout for all clients, unless overridden
|
||||
;Default - 60 seconds
|
||||
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
|
||||
; when we're not on hold
|
||||
|
||||
;Type of Service
|
||||
;Default - none (lowdelay, thoughput, reliability, mincost, none)
|
||||
;tos=lowdelay
|
||||
|
||||
;amaflags = default
|
||||
|
||||
;The account code used by default for all clients.
|
||||
;accountcode=h3230101
|
||||
|
||||
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
|
||||
;Default - ulaw
|
||||
; ONLY ulaw, gsm, g729 and g7231 supported as of now
|
||||
disallow=all ;Note order of disallow/allow is important.
|
||||
allow=gsm
|
||||
allow=ulaw
|
||||
|
||||
|
||||
; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
|
||||
; h245alphanumeric, h245signal.
|
||||
;Default - rfc 2833
|
||||
dtmfmode=rfc2833
|
||||
;
|
||||
; round trip delay request, default = 0,0 (not send)
|
||||
; x - count of unreplied requests before hangup, y - interval in sec between requests
|
||||
;
|
||||
;roundtrip=x,y
|
||||
|
||||
;
|
||||
; FAX detection will cause the OOH323 channel to jump to the 'fax' extension (if it exists)
|
||||
; based one or more events being detected. The events that can be detected are an incoming
|
||||
; CNG tone or an incoming T.38 RequestMode packet
|
||||
;
|
||||
; yes - enable both detection (CNG & T.38)
|
||||
; no - disable both
|
||||
; cng - enable CNG detection (default)
|
||||
; t38 - enable T.38 request detection
|
||||
;
|
||||
faxdetect = cng
|
||||
|
||||
; User/peer/friend definitions:
|
||||
; User config options Peer config options
|
||||
; ------------------ -------------------
|
||||
; context
|
||||
; disallow disallow
|
||||
; allow allow
|
||||
; accountcode accountcode
|
||||
; amaflags amaflags
|
||||
; dtmfmode dtmfmode
|
||||
; rtptimeout ip
|
||||
; port
|
||||
; h323id
|
||||
; email
|
||||
; url
|
||||
; e164
|
||||
; rtptimeout
|
||||
|
||||
;
|
||||
|
||||
;
|
||||
; direct rtp between two remote peers, disabled by default
|
||||
; can be specified globally or per peer/user section
|
||||
;
|
||||
directmedia=no
|
||||
;
|
||||
; early direct rtp (try to establish direct rtp before asnwer)
|
||||
; disabled by default, auto enabled by directmedia is enabled
|
||||
; can be disabled escpecially if directmedia is enabled.
|
||||
; can be specified globally or per peer/user section
|
||||
;
|
||||
;
|
||||
directrtpsetup=no
|
||||
|
||||
;Define users here
|
||||
;Section header is extension
|
||||
[myuser1]
|
||||
type=user
|
||||
context=context1
|
||||
disallow=all
|
||||
allow=gsm
|
||||
allow=ulaw
|
||||
|
||||
|
||||
|
||||
[mypeer1]
|
||||
type=peer
|
||||
context=context2
|
||||
ip=a.b.c.d ; UPDATE with appropriate ip address
|
||||
port=1720 ; UPDATE with appropriate port
|
||||
e164=101
|
||||
|
||||
|
||||
|
||||
[myfriend1]
|
||||
type=friend
|
||||
context=default
|
||||
ip=10.0.0.82 ; UPDATE with appropriate ip address
|
||||
port=1820 ; UPDATE with appropriate port
|
||||
disallow=all
|
||||
allow=ulaw
|
||||
e164=12345
|
||||
rtptimeout=60
|
||||
dtmfmode=rfc2833
|
||||
117
asterisk/osp.conf
Executable file
117
asterisk/osp.conf
Executable file
@ -0,0 +1,117 @@
|
||||
;
|
||||
; Open Settlement Protocol Sample Configuration File
|
||||
;
|
||||
; This file contains configuration of OSP server providers that are used by the
|
||||
; Asterisk OSP module. The section "general" is reserved for global options.
|
||||
; All other sections describe specific OSP Providers. The provider "default"
|
||||
; is used when no provider is otherwise specified.
|
||||
;
|
||||
; The "servicepoint" and "source" parameters must be configured. For most
|
||||
; implementations the other parameters in this file can be left unchanged.
|
||||
;
|
||||
[general]
|
||||
;
|
||||
; Enable cryptographic acceleration hardware.
|
||||
; The default value is no.
|
||||
;
|
||||
;accelerate=no
|
||||
;
|
||||
; Enable security features.
|
||||
; If security features are disabled, Asterisk cannot validate signed tokens and
|
||||
; all certificate file name parameters are ignored.
|
||||
; The default value is no.
|
||||
;
|
||||
;securityfeatures=no
|
||||
;
|
||||
; Defines the status of tokens that Asterisk will validate.
|
||||
; 0 - signed tokens only
|
||||
; 1 - unsigned tokens only
|
||||
; 2 - both signed and unsigned
|
||||
; The default value is 0, i.e. the Asterisk will only validate signed tokens.
|
||||
; If securityfeatures are disabled, Asterisk cannot validate signed tokens.
|
||||
;
|
||||
;tokenformat=0
|
||||
;
|
||||
;[default]
|
||||
;
|
||||
; List all service points (OSP servers) for this provider.
|
||||
; Use either domain name or IP address. Most OSP servers use port 5045.
|
||||
;
|
||||
;servicepoint=http://osptestserver.transnexus.com:5045/osp
|
||||
;
|
||||
; Define the "source" device for requesting OSP authorization.
|
||||
; This value is usually the domain name or IP address of the the Asterisk server.
|
||||
;
|
||||
;source=domain name or [IP address in brackets]
|
||||
;
|
||||
; Define path and file name of crypto files.
|
||||
; The default path for crypto file is /var/lib/asterisk/keys. If no path is
|
||||
; defined, crypto files will in /var/lib/asterisk/keys directory.
|
||||
;
|
||||
; Specify the private key file name.
|
||||
; If this parameter is unspecified or not present, the default name will be the
|
||||
; osp.conf section name followed by "-privatekey.pem" (for example:
|
||||
; default-privatekey.pem)
|
||||
; If securityfeatures are disabled, this parameter is ignored.
|
||||
;
|
||||
;privatekey=pkey.pem
|
||||
;
|
||||
; Specify the local certificate file.
|
||||
; If this parameter is unspecified or not present, the default name will be the
|
||||
; osp.conf section name followed by "- localcert.pem " (for example:
|
||||
; default-localcert.pem)
|
||||
; If securityfeatures are disabled, this parameter is ignored.
|
||||
;
|
||||
;localcert=localcert.pem
|
||||
;
|
||||
; Specify one or more Certificate Authority key file names. If none are listed,
|
||||
; a single Certificate Authority key file name is added with the default name of
|
||||
; the osp.conf section name followed by "-cacert_0.pem " (for example:
|
||||
; default-cacert_0.pem)
|
||||
; If securityfeatures are disabled, this parameter is ignored.
|
||||
;
|
||||
;cacert=cacert_0.pem
|
||||
;
|
||||
; Configure parameters for OSP communication between Asterisk OSP client and OSP
|
||||
; servers.
|
||||
;
|
||||
; maxconnections: Max number of simultaneous connections to the provider OSP
|
||||
; server (default=20)
|
||||
; retrydelay: Extra delay between retries (default=0)
|
||||
; retrylimit: Max number of retries before giving up (default=2)
|
||||
; timeout: Timeout for response in milliseconds (default=500)
|
||||
;
|
||||
;maxconnections=20
|
||||
;retrydelay=0
|
||||
;retrylimit=2
|
||||
;timeout=500
|
||||
;
|
||||
; Set the authentication policy.
|
||||
; 0 - NO - Accept all calls.
|
||||
; 1 - YES - Accept calls with valid token or no token. Block calls with
|
||||
; invalid token.
|
||||
; 2 - EXCLUSIVE - Accept calls with valid token. Block calls with invalid token
|
||||
; or no token.
|
||||
; Default is 1,
|
||||
; If securityfeatures are disabled, Asterisk cannot validate signed tokens.
|
||||
;
|
||||
;authpolicy=1
|
||||
;
|
||||
; Set the default destination protocol. The OSP module supports SIP, H323, and
|
||||
; IAX protocols. The default protocol is set to SIP.
|
||||
;
|
||||
;defaultprotocol=SIP
|
||||
;
|
||||
; Set the work mode.
|
||||
; 0 - Direct
|
||||
; 1 - Indirect
|
||||
; Default is 0,
|
||||
;
|
||||
;workmode=0
|
||||
;
|
||||
; Set the service type.
|
||||
; 0 - Normal voice service
|
||||
; 1 - Ported number query service
|
||||
; Default is 0,
|
||||
;
|
||||
;servicetype=0
|
||||
152
asterisk/oss.conf
Executable file
152
asterisk/oss.conf
Executable file
@ -0,0 +1,152 @@
|
||||
;
|
||||
; Automatically generated from ../channels/chan_oss.c
|
||||
;
|
||||
|
||||
[general]
|
||||
; General config options, with default values shown.
|
||||
; You should use one section per device, with [general] being used
|
||||
; for the first device and also as a template for other devices.
|
||||
;
|
||||
; All but 'debug' can go also in the device-specific sections.
|
||||
;
|
||||
; debug = 0x0 ; misc debug flags, default is 0
|
||||
|
||||
; Set the device to use for I/O
|
||||
; device = /dev/dsp
|
||||
|
||||
; Optional mixer command to run upon startup (e.g. to set
|
||||
; volume levels, mutes, etc.
|
||||
; mixer =
|
||||
|
||||
; Software mic volume booster (or attenuator), useful for sound
|
||||
; cards or microphones with poor sensitivity. The volume level
|
||||
; is in dB, ranging from -20.0 to +20.0
|
||||
; boost = n ; mic volume boost in dB
|
||||
|
||||
; Set the callerid for outgoing calls
|
||||
; callerid = John Doe <555-1234>
|
||||
|
||||
; autoanswer = no ; no autoanswer on call
|
||||
; autohangup = yes ; hangup when other party closes
|
||||
; extension = s ; default extension to call
|
||||
; context = default ; default context for outgoing calls
|
||||
; language = "" ; default language
|
||||
|
||||
; If you set overridecontext to 'yes', then the whole dial string
|
||||
; will be interpreted as an extension, which is extremely useful
|
||||
; to dial SIP, IAX and other extensions which use the '@' character.
|
||||
; The default is 'no' just for backward compatibility, but the
|
||||
; suggestion is to change it.
|
||||
; overridecontext = no ; if 'no', the last @ will start the context
|
||||
; if 'yes' the whole string is an extension.
|
||||
|
||||
; low level device parameters in case you have problems with the
|
||||
; device driver on your operating system. You should not touch these
|
||||
; unless you know what you are doing.
|
||||
; queuesize = 10 ; frames in device driver
|
||||
; frags = 8 ; argument to SETFRAGMENT
|
||||
|
||||
; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
|
||||
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
|
||||
; OSS channel. Defaults to "no". An enabled jitterbuffer will
|
||||
; be used only if the sending side can create and the receiving
|
||||
; side can not accept jitter. The OSS channel can't accept jitter,
|
||||
; thus an enabled jitterbuffer on the receive OSS side will always
|
||||
; be used if the sending side can create jitter.
|
||||
|
||||
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
|
||||
|
||||
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
|
||||
; resynchronized. Useful to improve the quality of the voice, with
|
||||
; big jumps in/broken timestamps, usually sent from exotic devices
|
||||
; and programs. Defaults to 1000.
|
||||
|
||||
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of an OSS
|
||||
; channel. Two implementations are currently available - "fixed"
|
||||
; (with size always equals to jbmax-size) and "adaptive" (with
|
||||
; variable size, actually the new jb of IAX2). Defaults to fixed.
|
||||
|
||||
; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
|
||||
; The option represents the number of milliseconds by which the new
|
||||
; jitter buffer will pad its size. the default is 40, so without
|
||||
; modification, the new jitter buffer will set its size to the jitter
|
||||
; value plus 40 milliseconds. increasing this value may help if your
|
||||
; network normally has low jitter, but occasionally has spikes.
|
||||
|
||||
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
|
||||
; ----------------------------------------------------------------------------------
|
||||
|
||||
; below is an entry for a second console channel
|
||||
; [card1]
|
||||
; device = /dev/dsp1 ; alternate device
|
||||
|
||||
; Below are the settings to support video. You can include them
|
||||
; in your general configuration as [general](+,video)
|
||||
; The parameters are all available through the CLI as "console name value"
|
||||
; Section names used here are only examples.
|
||||
|
||||
[my_video](!) ; you can just include in your config
|
||||
videodevice = /dev/video0 ; uses your V4L webcam as video source
|
||||
videodevice = X11 ; X11 grabber. Dragging on the local display moves the origin.
|
||||
videocodec = h263 ; also h261, h263p, h264, mpeg4, ...
|
||||
|
||||
; video_size is the geometry used by the encoder.
|
||||
; Depending on the codec your choice is restricted.
|
||||
video_size = 352x288 ; the format WIDTHxHEIGHT is also ok
|
||||
video_size = cif ; sqcif, qcif, cif, qvga, vga, ...
|
||||
|
||||
; You can also set the geometry used for the camera, local display and remote display.
|
||||
; The local window is on the right, the remote window is on the left.
|
||||
; Right clicking with the mouse on a video window increases the size,
|
||||
; center-clicking reduces the size.
|
||||
camera_size = cif
|
||||
remote_size = cif
|
||||
local_size = qcif
|
||||
|
||||
bitrate = 60000 ; rate told to ffmpeg.
|
||||
fps = 5 ; frames per second from the source.
|
||||
; qmin = 3 ; quantizer value passed to the encoder.
|
||||
|
||||
; The keypad is made of an image (in any format supported by SDL_image)
|
||||
; and some configuration entries indicating the location and function of buttons.
|
||||
; These entries can also be contained in the comment field of the image,
|
||||
; which is a lot more convenient to manage.
|
||||
; E.g. for jpeg you can write them with wrjpgcom (part of libjpeg).
|
||||
; The format to define keys is
|
||||
; region = <event> <shape> x0 y0 x1 y1 h
|
||||
; where <event> is the event to be generated (a digit, pickup, hangup,...)
|
||||
; <shape> is the shape of the region (currently 'rect' and 'circle' are
|
||||
; supported, the latter is really an ellipse), x0 y0 x1 y1 are the
|
||||
; coordinates of the base of the rectangle or main diameter of the ellipse,
|
||||
; (they can be rotated) while h is the height of the rectangle or the other
|
||||
; diameter of the ellipse.
|
||||
;
|
||||
[my_skin](!)
|
||||
keypad = /tmp/keypad.jpg
|
||||
region = 1 rect 19 18 67 18 28
|
||||
region = 2 rect 84 18 133 18 28
|
||||
region = 3 rect 152 18 201 18 28
|
||||
region = 4 rect 19 60 67 60 28
|
||||
region = 5 rect 84 60 133 60 28
|
||||
region = 6 rect 152 60 201 60 28
|
||||
region = 7 rect 19 103 67 103 28
|
||||
region = 8 rect 84 103 133 103 28
|
||||
region = 9 rect 152 103 201 103 28
|
||||
region = * rect 19 146 67 146 28
|
||||
region = 0 rect 84 146 133 146 28
|
||||
region = # rect 152 146 201 146 28
|
||||
region = pickup rect 229 15 267 15 40
|
||||
region = hangup rect 230 66 270 64 40
|
||||
region = mute circle 232 141 264 141 33
|
||||
region = sendvideo circle 235 185 266 185 33
|
||||
region = autoanswer rect 228 212 275 212 50
|
||||
|
||||
; another skin with entries for the keypad and a small font
|
||||
; to write to the message boards in the skin.
|
||||
[skin2](!)
|
||||
keypad = /tmp/kpad2.jpg
|
||||
keypad_font = /tmp/font.png
|
||||
|
||||
; to add video support, uncomment this and remember to install
|
||||
; the keypad and keypad_font files to the right place
|
||||
; [general](+,my_video,skin2)
|
||||
51
asterisk/phone.conf
Executable file
51
asterisk/phone.conf
Executable file
@ -0,0 +1,51 @@
|
||||
;
|
||||
; Linux Telephony Interface
|
||||
;
|
||||
; Configuration file
|
||||
;
|
||||
[interfaces]
|
||||
;
|
||||
; Select a mode, either the phone jack provides dialtone, reads digits,
|
||||
; then starts PBX with the given extension (dialtone mode), or
|
||||
; immediately provides the PBX without reading any digits or providing
|
||||
; any dialtone (this is the immediate mode, the default). Also, you
|
||||
; can set the mode to "fxo" if you have a linejack to make it operate
|
||||
; properly. If you are using a Sigma Designs board you may set this to
|
||||
; "sig".
|
||||
;
|
||||
mode=immediate
|
||||
;mode=dialtone
|
||||
;mode=fxo
|
||||
;mode=sig
|
||||
;
|
||||
; You can decide which format to use by default, "g723.1", "g729", or "slinear".
|
||||
; Note that g729 is only supported for Sigma Designs boards.
|
||||
; XXX Be careful, sometimes the card causes kernel panics when running
|
||||
; in signed linear mode for some reason... XXX
|
||||
;
|
||||
format=slinear
|
||||
;format=g723.1
|
||||
;format=g729
|
||||
;
|
||||
; And set the echo cancellation to "off", "low", "medium", and "high".
|
||||
; This is not supported on all phones.
|
||||
;
|
||||
echocancel=medium
|
||||
;
|
||||
; You can optionally use VAD/CNG silence suppression
|
||||
;
|
||||
;silencesupression=yes
|
||||
;
|
||||
; List all devices we can use. Contexts may also be specified
|
||||
;
|
||||
;context=local
|
||||
;
|
||||
; You can set txgain and rxgain for each device in the same way as context.
|
||||
; If you want to change default gain value (1.0 =~ 100%) for device, simple
|
||||
; add txgain or rxgain line before device line. But remember, if you change
|
||||
; volume all cards listed below will be affected by these values. You can
|
||||
; use float values (1.0, 0.5, 2.0) or percentage values (100%, 150%, 50%).
|
||||
;
|
||||
;txgain=100%
|
||||
;rxgain=1.0
|
||||
;device => /dev/phone0
|
||||
142
asterisk/phoneprov.conf
Executable file
142
asterisk/phoneprov.conf
Executable file
@ -0,0 +1,142 @@
|
||||
[general]
|
||||
; This section applies only to the default sip.conf/users.conf config provider
|
||||
; embedded in res_phoneprov. Other providers may provide their own default settings.
|
||||
|
||||
; The default behavior of res_phoneprov will be to set the SERVER template variable to
|
||||
; the IP address that the phone uses to contact the provisioning server and the
|
||||
; SERVER_PORT variable to the bindport setting in sip.conf. Unless you have a very
|
||||
; unusual setup, you should not need to set serveraddr, serveriface, or serverport.
|
||||
|
||||
;serveraddr=192.168.1.1 ; Override address to send to the phone to use as server address.
|
||||
;serveriface=eth0 ; Same as above, except an ethernet interface.
|
||||
; Useful for when the interface uses DHCP and the asterisk http
|
||||
; server listens on a different IP than chan_sip.
|
||||
;serverport=5060 ; Override port to send to the phone to use as server port.
|
||||
default_profile=polycom ; The default profile to use if none specified in users.conf
|
||||
|
||||
; You can define profiles for different phones specifying what files to register
|
||||
; with the provisioning server. You can define either static files, or dynamically
|
||||
; generated files that can have dynamic names and point to templates that variables
|
||||
; can be substituted into. You can also set arbitrary variables for the profiles
|
||||
; templates to have access to. Profiles are shared across all config providers.
|
||||
; Example:
|
||||
|
||||
;[example]
|
||||
;mime_type => application/octet-stream
|
||||
;static_file => example/firmware
|
||||
;static_file => example/default.cfg,text/xml
|
||||
;${TOUPPER(${MAC})}.cfg => templates/example-mac.cfg
|
||||
;setvar => DB_CIDNAME=${ODBC_CID_NAME_LOOKUP(${USERNAME})}
|
||||
|
||||
; Dynamically generated files have a filename registered with variable substitution
|
||||
; with variables obtained from various config providers. The default provider
|
||||
; embedded in res_phoneprov reads users.conf. Other providers will have their own
|
||||
; sources for the variables and may provide additional variables not listed here.
|
||||
|
||||
; Built in variables and the options in users.conf that they come from
|
||||
; MAC (macaddress)
|
||||
; USERNAME (username)
|
||||
; DISPLAY_NAME (fullname)
|
||||
; SECRET (secret)
|
||||
; LABEL (label)
|
||||
; CALLERID (cid_number)
|
||||
; VOCIEMAIL_EXTEN (vmexten)
|
||||
; EXTENSION_LENGTH (localextenlength)
|
||||
; LINE
|
||||
; LINEKEYS
|
||||
|
||||
; Built-in variables and the options in phoneprov.conf that they come from
|
||||
; SERVER (server)
|
||||
; SERVER_PORT (serverport)
|
||||
|
||||
|
||||
; Built-in variables for managing timezones and daylight savings time.
|
||||
; TZOFFSET
|
||||
; DST_ENABLE
|
||||
; DST_START_MONTH
|
||||
; DST_START_MDAY
|
||||
; DST_START_HOUR
|
||||
; DST_END_MONTH
|
||||
; DST_END_MDAY
|
||||
; DST_END_HOUR
|
||||
; TIMEZONE
|
||||
|
||||
[polycom]
|
||||
staticdir => configs/ ; Sub directory of AST_DATA_DIR/phoneprov that static files reside
|
||||
; in. This allows a request to /phoneprov/sip.cfg to pull the file
|
||||
; from /phoneprov/configs/sip.cfg
|
||||
mime_type => text/xml ; Default mime type to use if one isn't specified or the
|
||||
; extension isn't recognized
|
||||
static_file => bootrom.ld,application/octet-stream ; Static files the phone will download
|
||||
static_file => bootrom.ver,plain/text ; static_file => filename,mime-type
|
||||
static_file => sip.ld,application/octet-stream
|
||||
static_file => sip.ver,plain/text
|
||||
static_file => sip.cfg
|
||||
static_file => custom.cfg
|
||||
static_file => 2201-06642-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2201-06642-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-11000-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11300-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11300-010.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11300-010.sip.ld,application/octet-stream
|
||||
static_file => 2345-11402-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11402-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-11500-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11500-010.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11500-020.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11500-030.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11500-030.sip.ld,application/octet-stream
|
||||
static_file => 2345-11500-040.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11500-040.sip.ld,application/octet-stream
|
||||
static_file => 2345-11600-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11600-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-11605-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-11605-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12200-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12200-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12200-002.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12200-002.sip.ld,application/octet-stream
|
||||
static_file => 2345-12200-004.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12200-004.sip.ld,application/octet-stream
|
||||
static_file => 2345-12200-005.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12200-005.sip.ld,application/octet-stream
|
||||
static_file => 2345-12365-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12365-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12500-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12500-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12560-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12560-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12600-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12600-001.sip.ld,application/octet-stream
|
||||
static_file => 2345-12670-001.bootrom.ld,application/octet-stream
|
||||
static_file => 2345-12670-001.sip.ld,application/octet-stream
|
||||
static_file => 3111-15600-001.bootrom.ld,application/octet-stream
|
||||
static_file => 3111-15600-001.sip.ld,application/octet-stream
|
||||
static_file => 3111-40000-001.bootrom.ld,application/octet-stream
|
||||
static_file => 3111-40000-001.sip.ld,application/octet-stream
|
||||
static_file => SoundPointIPWelcome.wav,application/octet-stream
|
||||
static_file => SoundPointIPLocalization/Japanese_Japan/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Norwegian_Norway/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Spanish_Spain/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Portuguese_Portugal/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/English_United_Kingdom/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/English_United_States/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Russian_Russia/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Italian_Italy/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Chinese_China/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Swedish_Sweden/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/English_Canada/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/German_Germany/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/French_France/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Danish_Denmark/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Dutch_Netherlands/SoundPointIP-dictionary.xml
|
||||
static_file => SoundPointIPLocalization/Korean_Korea/SoundPointIP-dictionary.xml
|
||||
|
||||
${MAC}.cfg => 000000000000.cfg ; Dynamically generated files.
|
||||
${MAC}-phone.cfg => 000000000000-phone.cfg ; (relative to AST_DATA_DIR/phoneprov)
|
||||
config/${MAC} => polycom.xml ; Dynamic Filename => template file
|
||||
${MAC}-directory.xml => 000000000000-directory.xml
|
||||
setvar => CUSTOM_CONFIG=/var/lib/asterisk/phoneprov/configs/custom.cfg ; Custom variable
|
||||
|
||||
;snom 300, 320, 360, 370, 820, 821, 870 support
|
||||
snom-${MAC}.xml => snom-mac.xml
|
||||
56
asterisk/pjproject.conf
Executable file
56
asterisk/pjproject.conf
Executable file
@ -0,0 +1,56 @@
|
||||
; Common pjproject options
|
||||
;
|
||||
|
||||
;[startup]
|
||||
; NOTES: The name of this section in the pjproject.conf configuration file must
|
||||
; remain startup or the configuration will not be applied.
|
||||
;
|
||||
;cache_pools = yes ; Cache pjproject memory pools for performance
|
||||
; Disable this option to help track down pool content
|
||||
; mismanagement when using valgrind or MALLOC_DEBUG.
|
||||
; The cache gets in the way of determining if the
|
||||
; pool contents are used after being freed and who
|
||||
; freed it.
|
||||
; Default yes
|
||||
;log_level=default ; Initial maximum pjproject logging level to log
|
||||
; Valid values are: 0-6, and default
|
||||
;
|
||||
; Note: This option is needed very early in the startup
|
||||
; process so it can only be read from config files because
|
||||
; the modules for other methods have not been loaded yet.
|
||||
;type= ; Must be of type startup (default: "")
|
||||
|
||||
;========================LOG_MAPPINGS SECTION OPTIONS===============================
|
||||
;[log_mappings]
|
||||
; SYNOPSIS: Provides pjproject to Asterisk log level mappings.
|
||||
; NOTES: The name of this section in the pjproject.conf configuration file must
|
||||
; remain log_mappings or the configuration will not be applied.
|
||||
; The defaults mentioned below only apply if this file or the 'log_mappings'
|
||||
; object can't be found. If the object is found, there are no defaults. If
|
||||
; you don't specify an entry, nothing will be logged for that level.
|
||||
;
|
||||
; These logging level meanings are typically used by pjproject:
|
||||
; - 0: fatal error
|
||||
; - 1: error
|
||||
; - 2: warning
|
||||
; - 3: info
|
||||
; - 4: debug
|
||||
; - 5: trace
|
||||
; - 6: more detailed trace
|
||||
;
|
||||
;asterisk_error = ; A comma separated list of pjproject log levels to map to
|
||||
; Asterisk errors.
|
||||
; (default: "0,1")
|
||||
;asterisk_warning = ; A comma separated list of pjproject log levels to map to
|
||||
; Asterisk warnings.
|
||||
; (default: "2")
|
||||
;asterisk_notice = ; A comma separated list of pjproject log levels to map to
|
||||
; Asterisk notices.
|
||||
; (default: "")
|
||||
;asterisk_verbose = ; A comma separated list of pjproject log levels to map to
|
||||
; Asterisk verbose.
|
||||
; (default: "")
|
||||
;asterisk_debug = ; A comma separated list of pjproject log levels to map to
|
||||
; Asterisk debug
|
||||
; (default: "3,4,5,6")
|
||||
;type= ; Must be of type log_mappings (default: "")
|
||||
Some files were not shown because too many files have changed in this diff Show More
Loading…
x
Reference in New Issue
Block a user