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

[Anthony Holloway]

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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