Problems with call answering on SIP trunk

[nirkkone]

Hi Everyone.

I have been trying to get 3cx working with TeliaSonera SIP Trunk and have run into a couple of problems.

Calls work fine when calling outbounds from 3cx, but with incoming calls there seems to be some oddities with SIP signalling:

When trying to answer the call 3cx sends m: audio 0 RTP/AVP back in the 200 OK message towards the trunk which means it does not declare any working codec which causes the TRUNK to discard the call. I noticed that in SDP of the SIP INVITE from the trunk the DynamicRTP-Type is 100 and repectively in a INVITE from 3cx towards the trunk the aforementioned value is 101. Could it be that 3cx just simply does not understand the INVITE if DynamicRTP-type is something else than 101???

Any help regarding this?

3cx version is 12 with the latest service packs installed.

 

[nirkkone]

Re: Problems with call answering and rejectiong on SIP trunk

Oh, and the SIP switch on the Trunk end sseems to be Siemens hiq9200.

 

[leejor]

It might help if you made mention of the actual VoIP provider. Another member may have had experience with them.

 

[nirkkone]

Sorry for that. The trunk provider is TeliaSonera.

 

[nirkkone]

OK. So. I have managed to get the calls through by disablin the PBX Delivers Audio setting in the provider options. This however causes the RTP traffic from the Provider to get stuck in our firewall. RTP ports are of course forwarded only to the 3cx server. Thus voice goes only one direction (outbounds)

If I change the settings of the IP phone to offer the same payload type for DTMF as the provider (100) calls work perfectly inside the 3cx PS. Anyhow, for some reason when a INVITE comes from the Trunk with 100 as the DTMF payload, 3cx answers with no matching codec in the SDP header.

Anyone have any idea is the payload type 100 statically defined for some other use in 3cx in regard of Trunks? I asked the provider and they have no possibility to change it since the change would affect their whole network.