[Spce-user] i can't pass dtmf
agranig at sipwise.com
Mon Jan 10 15:57:18 EST 2011
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
On 01/10/2011 03:35 PM, pablo umanzor wrote:
> so, the flow of call is
> 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
> 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,
>> 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 126.96.36.199:5060 -> 188.8.131.52:5060
>>> SIP/2.0 404 Not Found'
>>> Via: SIP/2.0/UDP 184.108.40.206:5060;branch=0'
>>> To: <sip:220.127.116.11>;tag=74cv5lgvhlhc6lafqu3m'
>>> From: <sip:pinger at sipwise.local>;tag=915eed9'
>>> Call-ID: 6db69307-38595286-945003 at 18.104.22.168'
>>> CSeq: 1 OPTIONS'
>>> Content-Length: 0'
>>> where i can get more information about this issue into logs?
>>> pablo umanzor
>>> Spce-user mailing list
>>> Spce-user at lists.sipwise.com
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
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