Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's.
Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the phones, I changed the DTMF mode to "rfc2833". The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the "right" way to fix this, as Polycom's configurations, by default, will "encode DTMF in the active RTP stream". There may have been a change in the sip channel's code that is causing this. Others on the list have indicated that they worked around the problem by reverting the version of the sip app to an older version. As the new code usually fixes other problems, the solution of reverting seemed to be counter-productive, so I tried other DTMF signalling modes. Thankfully, the stock Polycom configs will work with Asterisk's sip.conf "rfc2833" DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread "*-1.0.7 DTFM => Not working" from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. Sincerely, David Gomillion _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users