08-29-2008 12:25 AM - edited 03-15-2019 12:55 PM
Hi,
I have a problem with my SIP trunk to my voice carrier. I have CUCM6 with a SIP trunk to a CUBE and then a SIP trunk from the CUBE to my carrier.
When I make a call from an internal phone to a PSTN number I see a SIP INVITE hit the CUBE but the content length is 0 and there is no SDP information.
Is there a command or parameter tweak on CUCM6/CUBE to enable this?
My internal IP phones are running a SCCP image.
Any help would be highly appreciated.
PS. This is a new system and has never worked.
08-29-2008 03:00 AM
It looks like the SDP information was being dropped due to G729 being set in the region configuration.
This has been changed to G711 and MTP is enabled on the SIP trunk.
SDP is now being sent but there is no codec specified?
See debug output below on a outbound to PSTN call:
Aug 29 10:50:40.775: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:PSTN_NUMBER@CUBE:5060 SIP/2.0
Date: Fri, 29 Aug 2008 10:50:40 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: <8599>;tag=ec6acf8b-8e42-492f-aa9a-7046172ba6ff-478518828599>
Allow-Events: presence, kpml
Supported: timer,replaces
Min-SE: 1800
Remote-Party-ID: <8599>;party=calling;screen=yes;privacy=off8599>
Content-Length: 216
User-Agent: Cisco-CUCM6.1
To:
Contact: <8599>8599>
Expires: 180
Content-Type: application/sdp
Call-ID: 50b68580-8b71d480-5a-f2911fac@CUCM6
Via: SIP/2.0/TCP CUCM6:5060;branch=z9hG4bK6411c183e2
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 70
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 CUCM6
s=SIP Call
c=IN IP4 CUBE
t=0 0
m=audio 16630 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Aug 29 10:50:40.779: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/TCP CUCM6:5060;branch=z9hG4bK6411c183e2
From: <8599>;tag=ec6acf8b-8e42-492f-aa9a-7046172ba6ff-478518828599>
To:
Date: Fri, 29 Aug 2008 10:50:40 GMT
Call-ID: 50b68580-8b71d480-5a-f2911fac@CUCM6
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Content-Length: 0
08-29-2008 05:35 AM
The codec 0 (G711/PCMU) is specified. The INVITE from CUCM looks good. The question is why is the 488 being returned.
You may want to "deb ccsip messages" on the CUBE to see if it is sending anything to the carrier.
The cause=65 translates to "Bearer capability not implemented". Perhaps you have the CUBE configured to anchor media but do not have any DSPs installed or available??
-steve
08-29-2008 09:13 AM
Since the codec used in the trunk was changed to G.711, make sure the dial-peer accepting the call in CUBE supports that codec as well.
Here's a good integration guide:
Regards,
Michael.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide