[cisco-voip] DTMF SIP to Verizon, wrong payload type...
Jonathan Charles
jonvoip at gmail.com
Thu May 17 14:16:39 EDT 2012
I am a partner.
The Verizon config has a call policer on it that blocks all calls after 10.
On Thu, May 17, 2012 at 1:16 PM, Anthony Holloway <
avholloway+cisco-voip at gmail.com> wrote:
> Small trick I learned a few years back: remove the "partner" from the URL
> so non-partners can view it:
>
>
> http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>
> -Anthony
>
> 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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>>
>>
>
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