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

[Jonathan Charles]

I used that to configure the gateway originally... if you use it you will
break your network.

On Thu, May 17, 2012 at 1:07 PM, miken miken <miken at sisna.com> wrote:

> Configuration examples and explanations for all of the primary North
> America SIP providers can be found on this link. You need partner access to
> view it.
>
>
> http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>
> Thank you
> MikeN
>
> On Thu, May 17, 2012 at 11:01 AM, 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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