Unified CME: Problem with incoming call via SIP Trunk

[karpenko.p]

Hello everybody!

Brief information about equipment and setup: Cisco 2811, Unified CME, SIP Trunk for Incoming Calls, SIP Trunks for Outgoing Calls.

After upgrading from IOS version 15.1(1)T to 15.1(4)M3 we've got problem with incoming calls drop. Right after update procedure Toll Fraud Prevention mechanism was disabled. When incoming call received on phone, which number included in hunt group, It starts ringing but then immediately stops and begin showing information about missed call. This process tooks less than one second and after that everything starts from beginng: phone ringing, call dropped, phone ringing & etc.

Internal and Outgoing calls could be placed without any problems.

Please check attached configuration file and call logs. Debug information about successfull call collected before IOS upgrade.

Here is a quote from debug information:

Jan 24 21:02:31.798 MSK: //1259/005FEB389C00/SIP/Msg/ccsipDisplayMsg:
Sent: 
CANCEL sip:400@192.168.1.143:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bKAD18FD
From: "79184960590" <sip:79184960590@192.168.1.250>;tag=ADB9B4-F75
To: <sip:400@192.168.1.143>
Date: Tue, 24 Jan 2012 17:02:31 GMT
Call-ID: 9CE1838-45E411E1-8655D1DA-39D06BED@192.168.1.250
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1327424551
Reason: Q.850;cause=38
Content-Length: 0

I think this is main reason of behavior described above.

Any help will be appreciated!

Thank you.

[paolo bevilacqua]

Seems to me you're using a 7960 with SIP firmware.

You should not do that, because SIP phones do not work well with CME, and have many limitations.

Convert the 7960 to SCCP and everything should work well.

[karpenko.p]

Paolo, we're using different Linksys SPA IP phones. In this case we were using SPA962 for testing purposes.

[paolo bevilacqua]

Sorry, I did not read the trace correctly.

Can you try with a SCCP phone for best results.

[karpenko.p]

Thank you, Paolo.

[karpenko.p]

Is there any other causes of such behavior?

[ADAM CRISP]

Hi,

I've noticed the following,

1.

Sucsess trace: The Invite to your SIP phone contains the codecs send by the service provider (g711a&u)

Failure trace: The Invite to your SIP phone contains the codecs configured as voice-class codec 1

Quite why this would cause an issue at this stage I'm not sure, however you do not have a transcoder configured,so offering codecs to the SIP phone that are not available is likely to cause problems with codec mismatches.Maybe the new IOS is expecting a transcoder to be configured in this scenario.

Try configuring codec transparent on the SIP ephone (or create voice-class codec 3 containing only the transparent setting)

2. I'm not sure why you have global SIP redirection turned on, you could try turning this off unless it's needed.

3. This is unrelated to your problem, but I've noticed your service provider wants to use payload type 96 for DTMF. I've had problems with this payload type before, where 96 is already assigned internally to the cisco for a propriatory fax transport.This affects incoming calls, not outgoing. A workaround would be to assign the fax transport to something else. eg on your incoming dial-peer 1000, put something like "rtp payload-type cisco-codec-fax-ind 124".

good luck with this one.

Adam