SIP Routing in Yate
From Yate Documentation
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| − | From a routing module you can call an sip channel like this: | + | From a routing module (e.g regexroute.conf) you can call an sip channel like this: |
extension=sip/sip:user@ip:port | extension=sip/sip:user@ip:port | ||
| + | |||
| + | |||
| + | From accfile.conf you can configure a SIP line when Yate is registering itself to a remote server.<br> | ||
| + | Sections with enabled=no are always ignored so you have to changed it to enabled=yes. | ||
| + | |||
| + | Parameter description for SIP: | ||
| + | * protocol: The protocol to register on | ||
| + | * username: Account username | ||
| + | * password: Account password | ||
| + | * interval: Registration interval (in seconds) to request from server (ignored for jabber) | ||
| + | * description: Caller name to set on outgoing calls on this account if none specified when routing | ||
| + | * registrar: Registrar address. A port can be specified here (e.g. 1.2.3.4:7999) | ||
| + | * authname: Authentication name to use instead of username when authenticating | ||
| + | * domain: Authentication domain if any | ||
| + | * outbound: Optional outbound proxy server if different from registrar. A port can be specified here (e.g. 1.2.3.4:7999) | ||
| + | * ip_transport: Optional SIP transport to use (udp, tcp, tls). | ||
| + | * ip_transport_remoteip: IP address to connect to register the account. | ||
| + | * ip_transport_remoteport: IP port to connect to register the account. | ||
| + | * ip_transport_localip: UDP: This parameter is used in conjuction ip_transport_localport to identify the transport to use. | ||
| + | * TCP/TLS: Optional local ip to use when connecting ip_transport_localport: Local port. | ||
| + | * UDP only: This parameter is used to identify the transport to use | ||
| + | * localaddress: Local address to set in sent messages | ||
| + | ::- Set it to 'yes' to detect NAT and re-register with public ip when NAT is detected | ||
| + | ::- Set it to 'no' or ipaddress (e.g. 1.2.3.4 or 1.2.3.4:5060) to disable NAT detection | ||
| + | * keepalive: Optional interval for NAT keep alive. | ||
| + | '''Note:''' Default port is 5060 for udp/tcp and 5061 for tls | ||
| + | |||
| + | This is an example: | ||
| + | |||
| + | [test_sip] | ||
| + | enabled=yes | ||
| + | protocol=sip | ||
| + | username=me | ||
| + | description=Test SIP account | ||
| + | interval=600 | ||
| + | authname=metoo | ||
| + | password=1234 | ||
| + | domain=somewhere.org | ||
| + | registrar=10.0.0.1:5060 | ||
| + | outbound=10.0.0.1:5061 | ||
| + | localaddress=192.168.0.1:5062 | ||
Revision as of 17:07, 25 October 2012
From a routing module (e.g regexroute.conf) you can call an sip channel like this:
extension=sip/sip:user@ip:port
From accfile.conf you can configure a SIP line when Yate is registering itself to a remote server.
Sections with enabled=no are always ignored so you have to changed it to enabled=yes.
Parameter description for SIP:
- protocol: The protocol to register on
- username: Account username
- password: Account password
- interval: Registration interval (in seconds) to request from server (ignored for jabber)
- description: Caller name to set on outgoing calls on this account if none specified when routing
- registrar: Registrar address. A port can be specified here (e.g. 1.2.3.4:7999)
- authname: Authentication name to use instead of username when authenticating
- domain: Authentication domain if any
- outbound: Optional outbound proxy server if different from registrar. A port can be specified here (e.g. 1.2.3.4:7999)
- ip_transport: Optional SIP transport to use (udp, tcp, tls).
- ip_transport_remoteip: IP address to connect to register the account.
- ip_transport_remoteport: IP port to connect to register the account.
- ip_transport_localip: UDP: This parameter is used in conjuction ip_transport_localport to identify the transport to use.
- TCP/TLS: Optional local ip to use when connecting ip_transport_localport: Local port.
- UDP only: This parameter is used to identify the transport to use
- localaddress: Local address to set in sent messages
- - Set it to 'yes' to detect NAT and re-register with public ip when NAT is detected
- - Set it to 'no' or ipaddress (e.g. 1.2.3.4 or 1.2.3.4:5060) to disable NAT detection
- keepalive: Optional interval for NAT keep alive.
Note: Default port is 5060 for udp/tcp and 5061 for tls
This is an example:
[test_sip] enabled=yes protocol=sip username=me description=Test SIP account interval=600 authname=metoo password=1234 domain=somewhere.org registrar=10.0.0.1:5060 outbound=10.0.0.1:5061 localaddress=192.168.0.1:5062
