SIP Trunk Between CUCM and Asterisk

[kennedymacharia]

Hi All,

I have a trunk between cucm 11 and asterisk but when a call is made from asterisk to cucm it disconnects immediately it is picked. Any idea?

[Ayodeji Okanlawon]

That sounds like a media negotiation issue. We are going to need CUCM logs to know why. A couple of things to check. What is the codec that is advertised from Asterisk? What is the bit rate on the region that is configured between the SIP trunk and the endpoint in CUCM?

[kennedymacharia]

Hi,

Thanks for the reply. The logs are attached. The Asterisk IP is 10.5.10.6 and I am calling Extension 3015

[Ayodeji Okanlawon · Accepted solution]

Hi,

The issue you are having is the region config between the asterisk SIP trunk and cisco phones.

The region config is set to use 8kbps ( region default to JubileeTZ).  Your Asterisk server is only advertising G711, hence a xcoder is needed. Please change the region setting to use 64Kbps

++ CUCM logs +++

35738374.006 |09:32:08.637 |AppInfo  |DET-MediaManager-(166857)::preCheckCapabilities, region1=Default, region2=JubileeTZ, Pty1 capCount=2 (Cap,ptime)= (2,20) (4,20), Pty2 capCount=9 (Cap,ptime)= (90,20) (6,20) (4,20) (2,20) (86,20) (11,20) (12,20) (15,20) (16,20)
35738374.007 |09:32:08.637 |AppInfo  |DET-RegionsServer::matchCapabilities-- savedOption=0, PREF_NONE, regionA=(null) regionB=(null) latentCaps(A=0, B=0) kbps=8, capACount=2, capBCount=9
35738374.008 |09:32:08.637 |AppInfo  |RegionsServer: applyCodecFilterIfNeeded - no codecs remained after filtering so restored original 0 caps

---

35738374.009 |09:32:08.637 |AppInfo  |DET-MediaManager-(166857)::preCheckCapabilities, caps mismatch! Xcoder Reqd. kbps(8), filtered A[capCount=0 (Cap,ptime)=], B[capCount=5 (Cap,ptime)= (90,20) (11,20) (12,20) (15,20) (16,20)] allowMTP=0 numXcoderRequired=1 xcodingSide=1

++ CUCM tried to allocate a xcoder but didnt find any ++

5738376.003 |09:32:08.637 |AppInfo  |MRM::convertScmStringToStdString MRG-AmeyoQueue
35738376.004 |09:32:08.637 |AppInfo  |MRM::getXcodeDeviceGivenMrgl
35738376.005 |09:32:08.637 |AppInfo  |MRM::getXcodeDeviceGivenMrgl GETTING XCODE FROM DEFAULT LIST
35738376.006 |09:32:08.637 |AppInfo  |MediaResourceManager::sendAllocationResourceErr - ERROR - no transcoder device configured

[kennedymacharia]

Thanks man this fixed the issue. I could not see that. Thanks again

[Ratheesh Kumar]

Hi there

Can you post some debugs from the gateway here. Could you try the following,

1. Assign a Hardware MTP to the call
2. Assign software MTP to MRGL
3. Check MTP required on CUCM SIP Trunk

Hope this helps!

Cheers
Rath!

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[kennedymacharia]

hi,

I have already applied resources but the issue persist. I have attached some logs above

[Manish Gogna]

As Ayodeji suggested try changing the region setting between the IP Phones and SIP trunk to use 64kbps  ( to ensure that xcoder is not required ) and then test. If the issue persists then share the traces for one such failed call.

Manish

[Doan Khanh Tan Thanh]

please enable SIP debug log in CUCM and collect SDL log then share it here

you might refer to this document

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200787-How-to-Collect-Traces-for-CUCM-9-x-10-x.html

[AlexC00011]

has anyone figured out how to sip trunk connect asterisk to CUCM on a newer version by using pjsip.conf? 

if so, would you please share your config ? 

[dsalazar]

Yes, using cucm 11.5 and issabel pbx the trunk works bothways....