cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2841
Views
0
Helpful
3
Replies

CUBE - No SDP in SIP INVITE from CUCM6

dual_line
Level 1
Level 1

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.

3 Replies 3

dual_line
Level 1
Level 1

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-47851882

Allow-Events: presence, kpml

Supported: timer,replaces

Min-SE: 1800

Remote-Party-ID: <8599>;party=calling;screen=yes;privacy=off

Content-Length: 216

User-Agent: Cisco-CUCM6.1

To:

Contact: <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-47851882

To: ;tag=DD1C0BAC-DC1

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

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

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:

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a00808ead0f.shtml

Regards,

Michael.

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: