VoIP PSTN Gateway

Revision as of 14:46, 2 April 2013

Yate has the functionality of a Voip - PSTN gateway.

It's main task is to provide signailing interworking and to transform the information it receives on one side in information compatible with the other side, like SIP to ISDN and / or SIP to SS7.

VoIP <-> PSTN Gateway in Yate

Yate can connect the existing PBX to alternative VoIP providers providing cost savings for enterprises.

It can also be used by providers to connect their TDM network to IP networks providing cost savings for long distance calls.

For this application usage of Sangoma cards is necessary.

Configuration in Yate

To configure SIP to ISDN or SIP to SS7 (ISUP) you will need:

• signalling
• voice circuits that can be:

- local: Sangoma or Zaptel cards. This can be configurate in Yate in files: wpcard.conf and zapcard.conf.

- remote: MGCP (to control a media gateway) configured in mgcpca.conf

For configuring Yate as gateway from SIP to ISDN or SIP to SS7 (ISUP) you need to configure in ysigchan.conf file:

• an ISUP trunk is a container for an ISUP call controller
• an ISDN trunk is a container with a Q.931 call controller, a circuit group (voice circuits), a Q.921 data link.

Routing in Yate

The routing it is done from regexroute.conf in the same way for ISDN and/or ISUP by addressing the trunk configured in ysigchan.conf.

Syntax: sig/callednumber;trunk=trunkname

The callednumber is the called party's number. The trunk parameter is mandatory and indicates the trunk to use to make the call. trunkname must be the name of an active trunk configured in ysigchan.conf.

Parameters

There are some common parameters for both protocols like called number, caller number.

The circuit parameter has different behavior. For incoming calls in ISDN the circuit it can be modified, but for ISUP the circuit it cannot be modified.

For more information see the parameters described in ysigchan.conf file.

See also

• More About VoIP