SIP Routing in Yate

Latest revision as of 10:08, 23 November 2017

SIP routing in Yate can be done from a single routing module or from a combination of modules.

The calls can be routed directly to a SIP channel from a routing module (e.g regexroute.conf).

Another way to route the calls is by using a SIP line(gateway) configured in accfile.conf.

[edit] 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

[edit] Route to a SIP line

[edit] 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_localip / ip_transport_localport:

- UDP: Used to identify the listener to use

- TCP/TLS: Optional local ip/port to bind on when connecting

• 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

[edit] 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

• Routing in Yate
• Accfile module
• Regular expressions Route Module
• Telephony