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| | For this application, the usage of Sangoma cards is necessary. | | For this application, the usage of Sangoma cards is necessary. |
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| − | == Configuration in Yate ==
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| | | | |
| − | To configure SIP to ISDN or SIP to SS7 (ISUP) you will need:
| + | == Installing Sangoma cards under Linux== |
| − | * signaling - This is configured in [[Signaling module|ysigchan.conf]].
| + | |
| | | | |
| − | : For SIP-ISUP gateway configure:
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| − | : - an ISUP trunk (container for an ISUP call controller)
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| − | : or for SIP-ISDN gateway:
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| − | : - an ISDN trunk (container with a Q.931 call controller, a circuit group (voice circuits), a Q.921 data link and an HDLC signalling interface).
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| | | | |
| − | * voice circuits that can be:
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| − | :- local: Sangoma or Zaptel cards. This can be configurate in Yate in files: [[Wpcard| wpcard.conf]] and [[Zapcard|zapcard.conf]].
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| − | :- remote: MGCP (to control a media gateway) configured in [[MGCP_call_agent_module|mgcpca.conf]]
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| | | | |
| − | For more configuration example please see article [[Signaling module]].
| + | == Configuration in Yate == |
| − | | + | |
| − | == Routing in Yate== | + | |
| − | | + | |
| − | The routing it is done from [[Regular expressions|regexroute.conf]] in the same way for ISDN and ISUP, by addressing the trunk configured in ysigchan.conf.
| + | |
| − | | + | |
| − | ;sig/callednumber;trunk=trunkname
| + | |
| − | .*=sig/\0;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 [[Signaling module|ysigchan.conf]].
| + | |
| − | | + | |
| − | == Parameters ==
| + | |
| | | | |
| − | There are some common parameters for both protocols like called number, caller number.
| + | First step [[Installation |install Yate]]. |
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| − | The ''circuit'' parameter has different behavior. For incoming calls in ISDN the circuit can be modified, but for ISUP the circuit cannot be modified.
| + | |
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| − | For more information see the parameters described in [[Signaling module|ysigchan.conf]] file.
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| | '''See also''' | | '''See also''' |
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| − | * [[More About VoIP]] | + | * [[What is Yate]] |
Revision as of 17:52, 10 April 2013
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.
Yate as a VoIP - PSTN Gateway
Yate can connect an 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, the usage of Sangoma cards is necessary.
Installing Sangoma cards under Linux
Configuration in Yate
First step install Yate.
See also
