Calls Disconnected in SIP Trunk

[jonnel_ponesto]

Hi,

I'm having difficulties solving the problem in SIP Trunk. The client has a CISCO in Malaysia and OXE in HongKong both were connected by a SIP Trunk. The scenario is if there is an external caller going to CISCO and use the SIP trunk to call colleagues in Hongkong, the Alcatel set (IP Phone) rings but it disconnects during Off-hook. In its trace I found this message:

BYE sip:192.168.21.12:5060 SIP/2.0
Reason: Q.850;cause=47
Date: Wed, 08 Jun 2011 04:31:33 GMT
From: <sip:9830@10.1.254.37>;tag=bc94ee42-6452-4fac-8ac0-4a60f83055e4-24638600
P-Asserted-Identity: <sip:9830@10.1.254.37>
Content-Length: 0

But if internal users in CISCO tried to call the HK user, no issue at all. Also even if they are going to use the HK Trunks no problem. Any advice?

Thanks

[alex]

BYE sip:192.168.21.12:5060 SIP/2.0
Reason: Q.850;cause=47
Date: Wed, 08 Jun 2011 04:31:33 GMT
From: <sip:9830@10.1.254.37>;tag=bc94ee42-6452-4fac-8ac0-4a60f83055e4-24638600
P-Asserted-Identity: <sip:9830@10.1.254.37>
Content-Length: 0

But you spoiled it posting only this single message - who sent this? In which context?
Basically it means "resource unavailable".

[jonnel_ponesto]

Hi,

The complete trace is already attached. Cisco User is extension 9830 (Malaysia) and HK is 2229.

[alex]

The problem is on a Cisco side. It cannot handle G729 codec which is proposed by OXE.
Kick some asses of Cisco guys:
- they should configure that Cisco sends SDP in INVITE (now it is not sending SDP, it waits for the opposite side to announce its possibilities)
- Possibly CUCM lacks MTP - it is needed for transcoding in case different codecs are used.
- Regions (like IP domains in OXE) are not managed properly in CUCM - no support of G.729.

You also could try switch this IP domain to G711ulaw and check.

[jonnel_ponesto]

Hi Alex,

Appreciate you help. Thank you very much.