Inbound SIP calls not working

Unanswered Question
Dec 15th, 2009

Hey guys,

I hate to bother you with this again, but I'm having trouble with inbound SIP calls on new software (8.0.0).  Would anybody be willing to take a look at my config and make sure that I'm not missing something?  SIP trunk provider is NexVortex.  3 inbound DIDs (xxxxxx7206-8) directly route to blast group 501.  1 inbound DID (xxxxxx7209) routes to internal extension 301, and 1 inbound DID (xxxxxx7210) routes to internal extension 210.  I've gone back through my config multiple times in CCA, but am still hitting this problem.  Thanks in advance,

Seth

I have this problem too.
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Steven Smith Tue, 12/15/2009 - 11:31

There is a bug in the 15.0.1XA code with voice source groups. 

CSCtc86432

The workaround is to disable voice source groups. 

sethschmautz Tue, 12/15/2009 - 11:44

Maulik,

Are you saying that I have to revert to the old software?  I thought that this bug was fixed in 8.0.0?  New site.

Seth

P.S. This is on a UC520, not a UC540

Maulik Shah Tue, 12/15/2009 - 12:04

Yes - it will be in 8.0.1 - you can revert back to UC520 7.0.x SW pack unless you need a new feature in 8.0.0 in which case you need to add a workaround for the defect.

sethschmautz Tue, 12/15/2009 - 12:49

Hi Maulik,

Okay, I think that I have a working solution, but we're still just a little ways from where I would like to be.  By removing the call voice source groups, I have incoming and outgoing calls working properly.  However, it only seems to work on the G.711ulaw codec.  When I try to force the SIP trunk to use G.729r8, it breaks incoming audio on the telephone handsets.  Here are the commands that I am issuing:

SDPDUC_520(config)#voice class codec 1
SDPDUC_520(config-class)#no codec preference 1 g711ulaw

SDPDUC_520(config-class)#no codec preference 2 g729r8
SDPDUC_520(config-class)#codec preference 1 g729r8
SDPDUC_520(config-class)#codec preference 2 g711ulaw
SDPDUC_520(config-class)#exit
SDPDUC_520(config)#sip-ua
SDPDUC_520(config-sip-ua)#g729-annexb override
SDPDUC_520(config-sip-ua)#exit
SDPDUC_520(config)#exit

I remember that we needed to setup a transcoder so that this codec would work with AA and voicemail, do we need to do this for the handsets also?  This was the original article that you recommended at the time:

https://supportforums.cisco.com/docs/DOC-9562

Thanks for your help.

Seth

Maulik Shah Tue, 12/15/2009 - 15:30

The CLI looks ok - can you send me a log for a failed call on G729? You do not need a transcoder for calls to IP phone handsets but yes need it for voicemail / auto attendant.

sethschmautz Tue, 12/15/2009 - 16:44

Hi Maulik,

I'll send you the debug of the failed call tomorrow.  I have some training to complete in the morning, and then will be able to spend some more time with it in the afternoon.  Would you have some time in the afternoon to look at this with me via webex, email, or posting here?  Thanks,

Seth

Maulik Shah Tue, 12/15/2009 - 21:56

Am off site tomorrow so posting the results here would be best bet.

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