VoIP PSTN Gateway
Yate has the functionality of a VoIP - PSTN gateway.
It's main task is to provide signaling interworking and to transform the information it receives on one side in information compatible with the other side.
This article explains the functionality of Yate as SIP-ISDN or SIP-SS7 gateway.
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Yate as a VoIP - PSTN Gateway
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.
You can configure Yate as a gateway from SIP to ISDN or from SIP to SS7 (ISUP):
- using local circuits like Sangoma cards
File:Yate-VoIP-PSTN Gateway with local circuits.png
- using remote circuits like a MG (Media Gateway). Yate configured without RTP forward.
File:Yate-VoIP-PSTN Gateway with remote circuits without RTP.png
- using remote circuits like a MG (Media Gateway). Yate configured with RTP forward.
File:Yate-VoIP-PSTN Gateway with remote circuits with RTP.png
You can read more about how RTP forward works in Yate in a dedicated article.
Configuration in Yate
To configure SIP to ISDN or SIP to SS7 (ISUP) you will need:
- signaling
- 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 and an HDLC signalling interface.
For more configuration example please see article Signaling module.
Routing in Yate
The routing it is done from regexroute.conf in the same way for ISDN and ISUP, by addressing the trunk configured in ysigchan.conf.
;sig/callednumber;trunk=trunkname ^123$=sig/123456789;trunk=trunk1
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 can be modified, but for ISUP the circuit cannot be modified.
For more information see the parameters described in ysigchan.conf file.
See also
