Can't establish a Voice gateway (cisco 2911) using SIP with CUCM 9.1

[Minh Tran]

I have configured a Cisco 2911 as a Voice Gateway using SIP (the configuration is attached), but unfortunately can't establish a test call to a phone (CUPC 8.6 SCCP) using csim start. I have done logging the ccsip debug and ccapi debug and attached them. Could anyone help me to solve this problem?

[Jaime Valencia · Accepted solution]

Why don't you try an actual call??

[Minh Tran]

Hi Jaime,

Thank your reply, but i think csim is enough for my purpose: Testing the SIP Gateway integrate with CUCM.

[Sreekanth Narayanan · Accepted solution]

It's failed with cause 16, which is normal. 

What is the number that you are calling? Can you enable debug voip ccapi inout and debug ccsip info, debug ccsip error before you try the csim?

Also send the logs to the buffer, so you don't lose any messages.

*Apr 17 10:10:08.037: //39/49ABDC418074/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call

Here's a good tool on debugs:

http://www.cisco.com/web/tsweb/tools/voice/multiservicedebuglookup.html

hantale

Sree

[Minh Tran]

Dear Sreekanth,

I placed a csim test call to DN 2501, which is my CIPC. And i have attached the log which is logged the output of debug voice ccapi inout, debug ccsip info, debug ccsip error before i try the csim.

[Sreekanth Narayanan]

I just did some research on my end and csim is not supported for SIP. The Invite will never be created and sent to the CUCM to initate the call. It disconnects in the router itself with normal cause.

*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPIOutgoingCallSDP: 
 Could not create source SDP for Outgoing Call
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPICreateOutboundSDP: 
 Error in creating an SDP for the outbound call - Check for supported codecs
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/preprocessSetup: 
 Error during outbound SDP creation
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call

Please use an actual call to test your dial-peer and integration with call manager. csim will not work.

Hantale

Sree

[Minh Tran]

Thank Sreekanth, i did capture the packet egress from the router and found out that no INVITE, just like you did. I'll use an actual call to test the SIP gateway and SIP Trunk. Will update the result later.

[Minh Tran]

Thank Jaime and Sree for your answers.I did try an actual call and it worked. Now i can see that csim did not support SIP.