[cisco-voip] DTMF SIP to Verizon, wrong payload type...

[Jonathan Charles]

It is not.

Per Verizon tech:

     Octet1058 SIP Message Body: SDP

--------------------------------------------------------------------------------
     ........  Header Field           v=0
     ........                         o=CiscoSystemsSIP-GW-UserAgent 794
632 IN IP4 1,1,1,1
     ........                         s=SIP Call
     ........                         c=IN IP4 1.1.1.1
     ........                         t=0 0
     ........                         m=audio 17176 RTP/AVP 0 101
     ........                         c=IN IP4 1.1.1.1
     ........                         a=rtpmap:0 PCMU/8000
     ........                         a=rtpmap:101 X-NSE/8000   <-- should
be telephone-event/8000
     ........                         a=fmtp:101 192-194
     ........                         a=ptime:20

They say the problem is on our end, and since we are sending the wrong
DTMF, they are closing their ticket.




On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
avholloway+cisco-voip at gmail.com> wrote:

> I'm glad you posted that.
>
> The m= is the actual setting for that call.  The a= are the available
> settings.  And you can see in the m=, you have codec 0 (g711) and DTMF 101
> (telephony).
>
> This looks correct.
>
> -Anthony
>
>
> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>
>> Added it, no change.
>>
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>> s=SIP Call
>> c=IN IP4 1.1.1.1
>> t=0 0
>> m=audio 18130 RTP/AVP 0 101
>> c=IN IP4 157.130.97.178
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 X-NSE/8000               <------------- this needs to be
>> telephone-event/8000
>> a=fmtp:101 192-194
>> a=ptime:20
>>
>>
>>
>>
>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>> avholloway+cisco-voip at gmail.com> wrote:
>>
>>> I see you are setting EO = Forced on the CUBE, which the telco requires,
>>> but are you using EO on the SIP trunk form CUCM to the CUBE?  What is your
>>> DTMF Signaling Method set to on that Trunk?
>>>
>>> The only command I run which I can see is missing from your config is:
>>>
>>> voice service voip
>>>   dtmf-interworking rtp-nte
>>>
>>> But I'm not positive that's your problem.
>>>
>>> -Anthony
>>>
>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>>>
>>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>>> work fine, however, for toll free calls, DTMF does not function.
>>>>
>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>> this:
>>>>
>>>> a=rtpmap:101 X-NSE/8000
>>>>
>>>> And it should be:
>>>>
>>>> telephone-event/8000
>>>>
>>>> And that is why it is failing.
>>>>
>>>>
>>>>
>>>> What configuration change can we do to force it to send the right DTMF
>>>> method?
>>>>
>>>>
>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>> request), there is a software MTP and Transcoder on the router (both in
>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>
>>>> Relevant SIP Config:
>>>>
>>>>
>>>> !
>>>> voice call send-alert
>>>> voice rtp send-recv
>>>> !
>>>> voice service voip
>>>>  allow-connections h323 to h323
>>>>  allow-connections h323 to sip
>>>>  allow-connections sip to h323
>>>>  allow-connections sip to sip
>>>>  no supplementary-service sip refer
>>>>  redirect ip2ip
>>>>  h323
>>>>   h225 display-ie ccm-compatible
>>>>  modem passthrough nse payload-type 101 codec g711ulaw
>>>>  sip
>>>>   bind media source-interface MFR1
>>>>   early-offer forced
>>>>   midcall-signaling passthru
>>>> !
>>>> !
>>>>
>>>> dial-peer voice 800 voip
>>>>  description OUTBOUND Voice SIP calls to VzB
>>>>  destination-pattern 1800[2-9]......
>>>>  voice-class sip dtmf-relay force rtp-nte
>>>>  session protocol sipv2
>>>>  session target sip-server
>>>>  incoming called-number .
>>>>  dtmf-relay rtp-nte
>>>>  codec g711ulaw
>>>>  no vad
>>>>
>>>>
>>>> !
>>>> sip-ua
>>>>  retry invite 2
>>>>  retry bye 2
>>>>  retry cancel 2
>>>>  registrar dns:verizonsipgateway expires 3600
>>>>  sip-server dns:verizonsipgateway:5071
>>>>  g729-annexb override
>>>> !
>>>>
>>>>
>>>>  Jonathan
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>
>>
>
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