[Spce-user] i can't pass dtmf

[Andreas Granig]

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
>>>
>>> _______________________________________________
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>>> http://lists.sipwise.com/listinfo/spce-user
>>
>>
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>>

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