[Spce-user] i can't pass dtmf

pablo umanzor pabloumanzor at gmail.com
Mon Jan 10 09:35:29 EST 2011


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




More information about the Spce-user mailing list