[cisco-voip] DTMF SIP to Verizon, wrong payload type...
Nick Matthews
matthnick at gmail.com
Thu May 17 15:15:13 EDT 2012
They're actually right.
Remove this:
modem passthrough nse payload-type 101 codec g711ulaw
Or change it to something similar like:
modem passthrough nse payload-type 100 codec g711ulaw
-nick
On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <
avholloway+cisco-voip at gmail.com> wrote:
> Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
> setting in CUCM on the trunk? And lastly, your MTP Required check box
> setting on the trunk?
>
> -Anthony
>
>
> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip at gmail.com>wrote:
>
>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20120517/34b69793/attachment.html>
More information about the cisco-voip
mailing list