CUCM Calls to ITSP via CUBE

[btmulgrew]

Hi - We are setting up a CUBE with CUCM connected to our ITSP.  Inbound calls are working fine, however we are receiving a SIP 503 for outbound calls as below:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/TCP 172.30.2.33:5060;branch=z9hG4bK40fc99b489d

From: "Test Phone" <sip:ipphone@CUCMIP>;tag=28458~cf036381-34e6-41aa-bacf-018                                                                                        3204bae3d-40019195

To: <sip:testnumber@CUBE_LAN_IP>;tag=1AADF722-170E

Date: Tue, 07 Apr 2015 05:28:49 GMT

Our call flow is:

IP Phone > CUCM > CUBE > ITSP

I can see the CUBE inbound and outbound dial peers being matched correctly and signalling and media are bound to the internal and external interfaces on the CUBE and media is set to flow through.  My understanding is that the sent SIP INVITE from the CUBE would be to the ITSP IP, however,  it is sending to the internal CUBE LAN IP as per:

sip:testnumber@CUBE_LAN_IP

I thought of setting a SIP Profile to normalize these messages, but think I may just have mis-configured something?

Thanks

Brian

[Aaron Harrison]

Hi

You would not need a SIP profile to get calls routing correctly. It will almost definitely be a misconfig.

Can you post up a full debug ccsip messages trace and your config?

Aaron

[btmulgrew]

Thanks AAron - I have attached the config and debug.  Your thoughts would be appreciated.

Cheers

Brian

[Aaron Harrison]

Hi 

Can you;

- remove the sip-profile

- do another debug with debug voip dialpeer default and debug ccsip messages enabled for the same call?

Aaron

[btmulgrew]

Thanks Aaron from, unfortunately I won't be able to generate fresh debug still tomorrow.... Though I am pretty sure we got the same SIP invite without the SIP profile.

thanks again

Brian

[btmulgrew]

Hi - Just an update on this.  After discussion with the ITSP, we set all SIP signalling to UDP and port 5060 on the voip dial peers and have made some progress, however, we are now receiving a SIP 404 Not Found from the provider.  I am pretty sure this is their end, but we are currently at a stalemate.  I will post the progress on this to the community.

Thanks

[Aaron Harrison]

Hi

So 404 generally means the far end can't route to that number; check you are properly stripping any access digits etc. Debug ccsip messages, check what goes out in the INVITE.

Aaron

[btmulgrew]

another day another session with the ITSP ;-)... we have managed to resolve the SIP 404 by making some changes to the gateway as per:

https://tools.cisco.com/bugsearch/bug/CSCuq47742/?referring_site=bugquickviewredir

sip-ua 
connection-reuse

We are now, however, receiving a SIP 400 Bad Request from CUCM.....:

Received: 
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.30.2.30:5060;branch=z9hG4bK7735BE
From: <sip:xxxxxx@172.30.2.30>;tag=43E7997A-CE
To: <sip:xxxxxx@172.31.80.33>;tag=62552~cf036381-34e6-41aa-bacf-0183204bae3d-73525713
Date: Wed, 15 Apr 2015 05:39:33 GMT
Call-ID: A0D5343B-E26811E4-B509E748-1D7A1B55@172.30.2.30
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=88
Server: Cisco-CUCM10.5
Content-Length: 0 

[Aaron Harrison]

Hi

Can you post the INVITE and other parts of the trace?

Aaron

[btmulgrew]

Hi - Just a final update,  we had a SIP Trunk from CUCM to the CUBE tunneling QSIG messages as there were also 2 x E1 PBX Connections on the same gateway. The ITSP did not like any QSIG being embedded into the SIP messages.

Note:  We are now integrating the E1s via MGCP and SIP end to end for the ITSP.

Thanks for your help AAron.

[inderpreetsingh23]

Hi,

Please try this:

voice service voip

sip

bind all source-interface <interface ip>

[btmulgrew]

Thanks - but we need to set the binding on each dial peer for CUCM and the ITSP to accept the appropriate IP addresses.

Cheers

[inderpreetsingh23]

This binding is configured for all the dial peers since you are configuring it on global basis.

Thanks

[Filipe Tavares]

btmulgrew,

If you are using another SIP Trunks in your gateway, you shouldn't bind all the media/control globally.

Like you have on your config, specify the bind under the dial-peers to the ITSP:

You can use this command to do so:

dial-peer voice XXXX voip
 voice-class sip bind control source-interface INTERFACE
 voice-class sip bind media source-interface INTERFACE

XXXX as your outbound dial-peer
INTERFACE as the interface that contains the IP address that you want to send to the ITSP

Remove the "voice-class sip profiles 101" under the outbound dial-peer and please add the debug.

Thanks,

Regards,