SIP Routing in Yate
From Yate Documentation
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| − | This page describes how | + | This page describes how the routing is done in Yate using SIP. |
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| + | You can route calls directly to a SIP channel from a routing module. | ||
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| + | Another way to route the calls is by using a SIP line(gateway). | ||
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| + | == Route to a SIP channel== | ||
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From a [[Routing|routing module]] (e.g regexroute.conf) you can call an sip channel like this: | From a [[Routing|routing module]] (e.g regexroute.conf) you can call an sip channel like this: | ||
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^100$=sip/sip:user@ip:port | ^100$=sip/sip:user@ip:port | ||
| − | + | == Route to a SIP line== | |
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| + | === Configure a SIP line=== | ||
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From [[Accfile|accfile.conf]] you can configure a SIP line. In this case Yate is registers to a remote server.<br> | From [[Accfile|accfile.conf]] you can configure a SIP line. In this case Yate is registers to a remote server.<br> | ||
Sections with enabled=no are always ignored so you have to changed them to enabled=yes. | Sections with enabled=no are always ignored so you have to changed them to enabled=yes. | ||
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localaddress=192.168.0.1:5062 | localaddress=192.168.0.1:5062 | ||
| − | + | ===Route to the configured line=== | |
Route a call to the [[Accfile|SIP line]] configured above: | Route a call to the [[Accfile|SIP line]] configured above: | ||
Revision as of 12:41, 30 May 2013
This page describes how the routing is done in Yate using SIP.
You can route calls directly to a SIP channel from a routing module.
Another way to route the calls is by using a SIP line(gateway).
Contents |
Route to a SIP channel
From a routing module (e.g regexroute.conf) you can call an sip channel like this:
;extension match rule=sip/sip:user@ip:port ^100$=sip/sip:user@ip:port
Route to a SIP line
Configure a SIP line
From accfile.conf you can configure a SIP line. In this case Yate is registers to a remote server.
Sections with enabled=no are always ignored so you have to changed them 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
- 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.
- 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: Configure a SIP line called [test_sip].
accfile.conf
[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
Route to the configured line
Route a call to the SIP line configured above:
regexroute.conf
^100$=line/100;line=test_sip
- this rule will match the number called: 100 and will route the call through the gateway
- the line parameter is fixed.
- line=test_sip will tell Yate on which gateway/account to route the call.
See also
