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

Thu May 17 14:49:18 EDT 2012

Even that example shows:

a=rtpmap:101 telephone-event/8000

Whereas I am seeing

a=rtpmap:101 X-NSE/8000

A: Why?
B: How do I change it?

On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
avholloway+cisco-voip at gmail.com> wrote:

> Here is the VoE I was talking about:
>
> https://communities.cisco.com/docs/DOC-7823
>
> Look towards the top for "VoE - CUBE SIP Trunking"
>
> Then download the PDF, and goto page 90.  The page is also discuss in the
> Webex recording @ 1h 48m 55s.  For those who cannot see this, it says:
>
> “c” parameter identifies the IP
>> address (20.1.1.1) that the peer
>> device should send the media to
>>
>
>
>> “m” parameter identifies:
>> the type of call (audio)
>> port number for media (16950)
>> payload type for the 1st
>> preferred codec (18 for G729)
>> dtmf (101 for RFC2833)
>>
>
>
>> “a’” parameter identifies all the
>> codecs and other descriptors for this
>> call leg
>
>
> This VoE event is very informative.  Hope that helps.
>
> -Anthony
>
> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>
>> 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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