Cisco UBE hangs up calls immediately after being answered

[otn-cco]

Hello,

Been working on a Cisco 2811 setup as a UBE device to terminate a few different voice techs (PRI and SIP), and send them via SIP to an IP-PBX.

I'm running into problems where the CUBE will immediately disconnect a call when you lift a handset to answer it, and then immediately sends another invite.  I've been looking at various debug ccsip output but there's a lot of information in there, and nothing that really jumps out as being a problem.

The call flow from a packet capture goes something like this (4.8 is the IP-PBX and 4.12 is the CUBE);

U 172.22.4.12:59803 -> 172.22.4.8:5061

INVITE sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 100 Trying.

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 180 Ringing.

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 180 Ringing.

### Pick up handset to answer

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 200 OK.

U 172.22.4.12:59803 -> 172.22.4.8:5061

ACK sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.12:59803 -> 172.22.4.8:5061

BYE sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 200 OK.

U 172.22.4.12:59803 -> 172.22.4.8:5061

INVITE sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.8:5061 -> 172.22.4.12:5060

SIP/2.0 100 Trying.

And the process repeats until the call times out.

My dial-peer config is;

dial-peer voice 501 voip

 destination-pattern 617567.....

 voice-class codec 1

 session protocol sipv2

 session target ipv4:172.22.4.8:5061

 session transport udp

 dtmf-relay rtp-nte

 no vad

!

dial-peer voice 502 voip

 voice-class codec 1

 incoming called-number 617567.....

 dtmf-relay rtp-nte

 no vad

Looking for some assistance as to the correct ccsip debugs to enable to see exactly where the problem is with the CUBE.  Pretty sure it's negotiating media properly, this would've been the first thing I looked at - but unfamiliarity with Cisco debugs makes it all a bit more challenging.

Thanks in advance,

Brendan

[Terry Cheema]

Firstly explain the call flow for the calls that are failing for example: From PSTN>CUBE>CUCM(?)>Phone

Second make a test and call, turn on the debug ccsip messages and debug voip ccapi inout, capture and attach along with your CUBE config.

Last if you can move this post to IP telephony section you will get better response.

-Terry

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[otn-cco]

Hi Terry,

Thanks, I didn't realise I had posted in Video over IP...

The call flow is:- SIP ITSP -> CUBE 172.22.4.12 -> IP-PBX (Asterisk) 172.22.4.8 -> Handset (extn 246).

I've attached the debug trace that you asked for.  One thing I noticed, it seems the ITSP is offering G.722.1, and in a later INVITE also offers iLBC.  I don't think either of these codecs are supported, and I also have PCMU and PCMA forced on the outgoing dial-peer.  Could this be a codec or transcoding problem?

Looking forward to responses...

-Brendan