Public SIP to Genesys SIP server via OXE

huntsman · Post by **huntsman** » 04 Nov 2011 05:26

Hi,

Having an issue at present with a OXE connected to a public sip provider and a genesys sip server.

The solution implemented on site is OXE appliance server connected to a public SIP provider via an ext gateway. Connection to the public carrier is OK.

The OXE is then connected to a geneysis sip server via another ext gateway using a abc-f trunk group. Nothing special there either.

Problem:- Geneysis sip server pushing calls to us with '0' in front of number, ARS prefix '0' setup to deal with outgoing call. When call to sent to us, SIP traces are showing a 503 'Service Unavailable' message, and OXE gives a B Channel choice error, stating there is no B channel available.

Note, the only Alcatel components are the appliance servers. There is no GD/INTIP/GA.

I am thinking there is a codec problem, but we had the genesys engineers make sure only G711 is coming to us, and we have told the ext gateways to use the algorithm stated in the trunk group, being G711.

Do I still need the compressors if the calls from the sip server is in G711? I don't know whether the compressors would still be used if all codecs match.

Any info would be great,

thanks.

Petr Halberstat · Post by **Petr Halberstat** » 06 Dec 2016 08:00

The ask about Outbound call from SIP server Genesys via OXE pbx PSTN.
If this call initate SIP server Genesys to OXE pbx, after OXE offered in STATUS OK SIP message SDP without rtpmap:97 telephone-event /8000, only rtpmap G723,G729,G711.
If the the call initate OXE to SIP Server Genesys, after in INVITE message OXE generate SDP also with rtpmap: 97 telephone-event /8000.
Pleas, do you may send to me somewhere explain of this differents behaviour ?

dryhouse · Post by **dryhouse** » 07 Dec 2016 07:09

Hi huntsman,

you are managed the External Callback Translation??..DEF...

Regards.

Post by **cavagnaro** » 07 Dec 2016 07:26

The sip trunk has a public network category that allows all calls?

Enviado de meu E6633 usando Tapatalk

Barrykeery · Post by **Barrykeery** » 12 Jul 2019 01:14

So, if you are not routing calls within the SIP Server via the route it comes in and all you want to do is see the Caller ID of the incoming call then I believe that Sarond is on the right track.

Trunks / SIP / SIP Peer Profile / Calling Line ID: Public Calling Party Number Passthrough and Use Original Calling Party Number If Available

Thanks,

TE

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