Author Topic: SIP trunking with Alcatel  (Read 4246 times)

Offline martyn

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SIP trunking with Alcatel
« on: December 05, 2012, 01:02:57 AM »
Has anyone managed to get a SIP trunk working between a 3300 and an alcatel before? The 3300 is running MCD 5.

I have got the trunk established, and can call from the Mitel to the alcatel without any issues, two way voice is fine.
Coming from the alcatel to the mitel, the mitel phone rings, but there is no audio either way.

From a wireshark trace it seems that the RTP stream never actually tries to establish.


Offline matthew

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Re: SIP trunking with Alcatel
« Reply #1 on: December 05, 2012, 01:51:04 AM »
You don't mention what kind of Alcatel. If it's an OXO (OmniPCX Office), they hard set the framing value if a specific codec is chosen for the link. It's probably easiest for you to try changing things at the Mitel end to see if it helps. Alcatel were fond of setting the codec to G729-30. I've seen some higher due to trying to make inband DTMF signalling work. You might try setting the RTP Packetization Rate Override to yes, and the Rate to 30, 40, etc. It's possible for the Alcatel to have the codec/framing set to Default which lets the remote end negotiate the codec. They also have an option for Direct RTP which is off by default that might help. Other than that, I'm out of ideas. Their SIP configuration is fairly basic.

Offline martyn

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Re: SIP trunking with Alcatel
« Reply #2 on: December 05, 2012, 06:07:24 AM »
Apologies for the lack of info about the Alcatel, it's not a system that I've touched before, and not one that I technically have access to to troubleshot in this instance.

I believe it is an OXO, but will check that tomorrow. There are two limiting problems aside from not knowing the Alcatel. The first is the distance, it is from australia to Vietnam. The second is the language barrier and the experience of the tech over there!

I can get access to the system if required, but having never seen it before makes it harder, but I will check out your suggestions. From what I could see though yes the sip config options are very simple and limited.


 

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