You could also bypass pjsip entirely and just have your main loop poll your DTMF code directly, instead of using the capabilities infrastructure. You'll need locks to protect the dtmf event queue either way because the audio driver is a different thread than the application. Bill On 4/7/15 9:29 AM, dpn at switchvoice.com wrote: > Hi Bill, > Thank you. You help is really appreciated! > I will dig in a code to get better view of what you are saying. > Best Regards > Dimitar > *From:* Bill Gardner <mailto:billg at wavearts.com> > *Sent:* Tuesday, April 7, 2015 16:03 > *To:* pjsip at lists.pjsip.org <mailto:pjsip at lists.pjsip.org> > *Subject:* Re: [pjsip] Inband DTMF > Hi Dimitar, > > You could try to duplicate the method currently used to get dtmf > callbacks from incoming RTP, renaming things appropriately. > > Even simpler, but really a hack, would be to define a new audio device > capability that would let the main loop poll the audio device for DTMF > events using the pjsip audio capabilities infrastructure. You'd have > to locally queue the incoming DTMF events in your device and > periodically check them from your main loop using a timer event. There > isn't anything built in that lets the audio device generate an > application callback. > > Regards, > > Bill > > On 4/6/15 5:27 PM, dpn at switchvoice.com wrote: >> Hi Gents, >> I have added dtmf detection to my tdm audio device. >> I have used spandsp tdmf detection which appears to work fine for my >> application. >> Now I am wondering what would be the best way to pass the dialed >> extension from my audio_device to the pjsua application layer. >> The spandsp dtmf allows an on_dtmf() callback to be hooked so we can >> have notification on each digit. >> Any comments will be appreciated. >> Dimitar >> *From:* dpn at switchvoice.com <mailto:dpn at switchvoice.com> >> *Sent:* Saturday, April 4, 2015 03:18 >> *To:* pjsip list <mailto:pjsip at lists.pjsip.org> >> *Subject:* [pjsip] Inband DTMF >> Hi Gents, >> I am tiring to integrate analog phone thru SLIC from Silicon labs to >> the pjsip. >> Thanks to the help from the this mailing list I have managed to move >> the task >> up to the point I have clean audio between my analog phone and a sip >> client >> analog_phone - pjsip <---> (Asterisk SIP server) <---> SIP client >> My SLIC doesn?t have built in DTMF detection. >> I just read that pjmedia doesn?t support inband DTMF detection. >> Can you please confirm/refuse that ? >> If DTMF doesn?t supported I am willing to borrow the inband DTMF >> detection >> from Asterisk or some other open implementation and put it in pjmedia. >> Do you see benefits for the project? >> As I am very new to pjsip I am not sure which will be the native way >> to integrate such a detector in pjsip. >> Is someone willing to advice me on the best way to put DTMF detection >> in the pjmedia? >> Thanks >> Dimitar >> ------------------------------------------------------------------------ >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> >> _______________________________________________ >> Visit our blog:http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > ------------------------------------------------------------------------ > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150407/869b212a/attachment.html>
