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CVP CUBE DTMF not working

Mark Applebee
Beginner
Beginner

     CVP with CUBE, DTMF is not worknig for any calls. CVP Logs below:

4192: 172.17.19.101: Sep 19 2012 20:56:47.739 -0400: %CVP_9_0_IVR-7-CALL:  {Thrd=http-8000-Processor22} VBServlet:service: HTTP Request from 172.17.253.252: { CALL_ID=04A5D28000010000000000AC6F1311AC, MSG_TYPE=CALL_RESULT, CALL_SEQ_NUM=1, ERROR_CODE=NO_ENTRY(17) } 

4193: 172.17.19.101: Sep 19 2012 20:56:47.739 -0400: %CVP_9_0_IVR-3-CALL_ERROR:  RunScript Error from 172.17.253.252 [NO_ENTRY(17)] CALLGUID: 04A5D28000010000000000AC6F1311AC DNIS=12711111111110148 {VRUScriptName: 'GD,-10' ConfigParam: '1,4,Y,2,8,1,1,N,N,#'} [id:3023]

.....

All the dial peer on CUBE is using dtmf-relay rtp-nte.CUBE logs are attached here. , anyone have an idea of why the dtmf is not working

Thanks

18 Replies 18

Ahmed Khalefa
Beginner
Beginner

Do you have inside the ICM Script : " Input Type " node is set to "D" , i.e: DTMF ?

i see you are running a GD microapp ... please post your ICM Script here if possible .

Thanks a lot,

Ahmed Salah

I do have use.microapp.input_type set to "D". Please see the script below

Please try to change the If.CED="12" to If.CED=="12" , and try again ...

Thanks A lot,

Ahmed Salah

I tried and it doent work. I honestly dont think that the problem, I tried menu micro apps in other script and they are also not getting the DTMF. I could see  in CVP logs that CVP is not geting any digit from gateway. But all my gw dial-peers are configured with rtp-nte

Can you post the VXML Voice GW configurations please ?

Thanks A lot,

Ahmed Salah

                   Here it is.

i can see that you allowed only sip to sip connection on the vxmlgw , and you are trying from an ip phone registered with a cucm ver8.6 as stated , any chance that you tried to check the Media Termination Point Required on the sip trunk to the Voice Gateway ?

Nope, that doen't make any difference.

please add the kpml relay into the vru dial-peer ,

i can see the dtmf signalling :

c=IN IP4 10.235.131.148

t=0 0

m=audio 30488 RTP/AVP 0 101

c=IN IP4 10.235.131.148

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Ahmed-I believe  payload type 101 is for rtp-nte. Where you seeing kpml here.

Mark, i can see it in the 1st INVITE Message came to the VGW from the CUCM :

INVITE sip:121355014@10.235.131.149:5060 SIP/2.0

Via: SIP/2.0/TCP 172.17.19.111:5060;branch=z9hG4bK2a521f66b10

From: <1356607>;tag=4277~197f0523-8faa-4c0c-b0eb-8005e8410b37-38680479

To: <121355014>

Date: Thu, 20 Sep 2012 00:56:34 GMT

Call-ID: 4a5d280-5a169c2-b1-6f1311ac@172.17.19.111

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM8.6

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Call-Info: <172.17.19.111:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Cisco-Guid: 0077976192-0000065536-0000000172-1863520684

Session-Expires:  1800

P-Asserted-Identity: <1356607>

Remote-Party-ID: <1356607>;party=calling;screen=yes;privacy=off

Contact: <1356607>

Max-Forwards: 70

Content-Length: 0

please check below in SIP Standard RFC :

3.3.7. Allow-Events header usage

   The "Allow-Events" header, if present, includes a list of tokens
   which indicates the event packages supported by the client (if sent
   in a request) or server (if sent in a response).  In other words, a
   node sending an "Allow-Events" header is advertising that it can
   process SUBSCRIBE requests and generate NOTIFY requests for all of
   the event packages listed in that header.

   Any node implementing one or more event packages SHOULD include an
   appropriate "Allow-Events" header indicating all supported events in
   all methods which initiate dialogs and their responses (such as
   INVITE) and OPTIONS responses.

   This information is very useful, for example, in allowing user agents
   to render particular interface elements appropriately according to
   whether the events required to implement the features they represent
   are supported by the appropriate nodes.

Ok. But I have configured RFC 2833 on CUCM trunk to GW. Also I tried to configure dtmf-relay sip-kpml to the dial-peers, but sip-kpml is not an option under my dial-peers, my gateway ios is 15.1(4)M3 tho.

Can you try using H.323 instead of SIP between the VGW and the CUCM ?

Unfortunately I don't have the right to do that. DTMF is not working for CTI RP calls to UCCE(agent transfer type) as well. Is there anyway I can turn off kpml from cucm?

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