listuser at spamomania.co.uk
2010-Jan-11 16:26 UTC
[asterisk-users] Sipgate > DTMF not detected
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an extension on an ata or a softphone, it works flawlessly (I have no fxo, everything is SIP based). It's extremely frustrating and I would be grateful if anyone could offer some help troubleshooting and fixing this?
On 11 Jan 2010, at 16:26, listuser at spamomania.co.uk wrote:> This has made no difference. I've tried a range of settings (auto, > rfc2833,info) but no matter what, it plain refuses to pick up key > presses. > <snip> > It's extremely frustrating and I would be grateful if anyone could > offer > some help troubleshooting and fixing this?Try asking Sipgate what settings you should use? If they are sending it as audio, make sure you are using suitable codecs etc. Try SIP traces to see what you can see. Steve
listuser at spamomania.co.uk wrote:> I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to > recognize digits pressed on a keypad coming in from a Sipgate trunk. > > There answer was to set this: > dtmfmode=rfc2833 > > in the general section of sip.conf > > This has made no difference. I've tried a range of settings (auto, > rfc2833,info) but no matter what, it plain refuses to pick up key > presses. > > Locally, if I call from an extension on an ata or a softphone, it works > flawlessly (I have no fxo, everything is SIP based). > > It's extremely frustrating and I would be grateful if anyone could offer > some help troubleshooting and fixing this? > > >Hi, maybe your RTP stream is not getting through the asterisk box due to "canreinvite=yes" setting in your SIP profile? What is result of the following test in your dialplan? exten => 123,1,NoOp(***INCOMING CALL***) exten => 123,n,Set(CHANNEL(language)=en) exten => 123,n,Answer() exten => 123,n,Read(CONFNO,conf-getconfno,4) exten => 123,n,Playback(conf-enteringno) exten => 123,n,SayDigits(${CONFNO}) exten => 123,n,Hangup Cheers Joern