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Calling Over SIP Trunk - Call disconnects

GRANT3779
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Hi All,

When making a call from - Local site over a remote sip trunk - call flow below

Local Site (call manager)  - over VPLS to - Remote Site (Voice Gateway) (Sip trunk connected to Gateway).

The call hits the SIP trunk and dials ok, the called party hear the phone ring, when they pick up, there is just silence, then calling party hears a busy tone after a few seconds.

Does this sound like an MTP issue? Where should I be looking? This was all working ok and just stopped working suddently. Callers local to the SIP trunk (e.g on the remote site) can call over the SIP no problem.

9 Replies 9

Chris Deren
Hall of Fame
Hall of Fame

Sounds like a codec issue.  What is the region set to between the phone and the SIP trunk?

What codec is service provider set for? Remember service providers will allow one codes which you need to follow.

Post "debug ccsip messages"

Chris

Hi Chris,

I'm guessing I need to run this command, then make a test call? Pretty new to the troubleshooting aspect of Voice.

Can I just check that when making a call over the SIP, is the whole call continually managed by an MTP until disconnect? What part does the MTP play? Where does the transcoding take place?

Thanks

Yes to first question.

What are you using MTP for?  What version of CUCM are you running? Recent versions should not require MTPs for any call flows. Is the SIP trunk on CUCM configured to use MTP?

Chris

Ok you can do a call detail report CDR on the serviceability page. on the report, if  your calls are being transcoded, You will see the destination device as being the transcoder. Check from both sides of the call.

However, you may not see the CDR showing the call as hitting a transcoder when you are making a test call from Side ' A' which is currnetly not work. The reason I say this is because I agree with what Chris said about this being a Codec issue. In which case, a transcoder was likely never triggered or triggered properly.

You can also use real-time monitoring tool to check if a transcoder becomes active during a call but you will have to make sure that  the cluster is silent for an accurate report.

Hi Guys,

Issue was an ASA in between the sites. A simple inspect h323 h225...

Removed this and calls now work..

With regards to the CDR - how do I view calls through this?

Regarding MTP Chris, I thought that the call from Call Manager (Site A) over SIP trunk (Site B) would always be terminated and handled by an MTP until call end?

Sorry...Like i say, very new to voice!

Thanks

No, MTPs with SIP trunks would be needed under the following conditions:

1. Early to Delayed offer negotiation, but in your environment since you have CUBE this would never be needed, even if you have direct SIP trunk to CUCM early offer is supported as of version 8.5

2. DTMF conversion, not needed as long as your phones support RFC2833 and SIP provider is using RFC2833

HTH, please rate all useful posts!

Chris

Hi Chris,

When you say CUBE- I assume this is my router acting as a gateway where the SIP trunk terminates?

So MTP, would this be used in external calls from CUCM over PRI lines?

Cheers

  i have run this command debug ccsip messages,

Apr 17 17:23:06.968: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:0126612068@172.22.7.14:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.7.10:5060;branch=z9hG4bKec67a5017a65e

From: "Ofentse" <sip:0110632482@172.22.7.10>;tag=007711e2-ab1e-440d-91a6-cf77d768eb88-24836815

To: <sip:0126612068@172.22.7.14>;tag=473EA0-1D81

Date: Wed, 17 Apr 2013 17:23:15 GMT

Call-ID: 7b874900-16e1da83-4440c-a0716ac@172.22.7.10

User-Agent: Cisco-CUCM8.0

Max-Forwards: 70

CSeq: 102 BYE

Reason: Q.850;cause=47

BELOW IS THE LOG FROM THE SERVICE PROVIDER. THEY HAVE INFORMED ME THAT THEY RECEIVE BYE MESSAGE FROM MY VOICE GATEWAY.

    Date: Wed, 17 Apr 2013 17:24:30 GMT

    Max-Forwards: 70

    User-Agent: Cisco-SIPGateway/IOS-12.x

    Reason: Q.850;cause=16

    Content-Length: 0

I think its also codec Mismatch!  what do u think guys??

Thanks

well cause code 16 is normal call clearing which useless information. However, cause code 47 can be a stronge indication of codec issues.

try to issue the following command in your router

show voice dsp group all

why dont you also copy and past your configs here? Are your transcoders registered?

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