[Spce-user] i can't pass dtmf

[pablo umanzor]

Thanks Andreas, it was a definition with dtmfmode in my asterisk box ,
i normalized this parameter with dtmfmode=rfc2833 and now is working!

regards
pablo umanzor

2011/1/10 Andreas Granig <agranig at sipwise.com>:
> Probably the DTMF tone is not recognized on the PSTN side? Check what
> method for DTMF transport you're using in your UA (SIP INFO, rfc2833 or
> inband), and check if it's correctly converted on your Asterisk box.
>
> This is actually not a CE issue, since it transparently relays it
> end-to-end.
>
> Andreas
>
> On 01/10/2011 03:35 PM, pablo umanzor wrote:
>> so, the flow of call is
>>
>> UA-------sip:providerCE--------Asterisk(sip/tdm)-------------PSTN------IVR(bank
>> or anything)
>>
>> first i make call to "bank" from my UA , the call is answered and the
>> ivr asks about available options, then i dial the digit 1 for example
>> but nothing happens, my question is where i can find more information
>> about this issue into sip_providerce or what log file can give more
>> information.
>>
>> regards
>> pablo umanzor
>>
>>
>> 2011/1/10 Andreas Granig <agranig at sipwise.com>:
>>> From your description, I and dozens of others have no idea how your
>>> setup looks like. You probably want to go towards a commercial service?
>>>
>>> Best regards,
>>> Andreas
>>>
>>> On 01/10/2011 06:48 AM, pablo umanzor wrote:
>>>> Hi,while a call is processed by an ivr system i can't pass dtmf , in
>>>> kamailio.cfg nothing happen and the output with ngrep-sip is (while i
>>>> pressing a digit):
>>>>
>>>>
>>>> U 2011/01/10 02:36:47.110361 200.1.0.2:5060 -> 9.8.7.6:5060
>>>> SIP/2.0 404 Not Found'
>>>> Via: SIP/2.0/UDP 9.8.7.6:5060;branch=0'
>>>> To: <sip:200.1.0.2>;tag=74cv5lgvhlhc6lafqu3m'
>>>> From: <sip:pinger at sipwise.local>;tag=915eed9'
>>>> Call-ID: 6db69307-38595286-945003 at 9.8.7.6'
>>>> CSeq: 1 OPTIONS'
>>>> Content-Length: 0'
>>>> '
>>>> where i can get more information about this issue into logs?
>>>>
>>>> regards
>>>> pablo umanzor
>>>>
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>>>
>>>
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>
>