Thanks Noah for your time, I am using rfc2833 as dtmf mode
I already tweaked the dialplan.digitmap="" (to an empty string) so everything gets out. my phone's sip.cfg codec setting looks like <preferences voice.codecPref.G711Mu="1" voice.codecPref.G711A="2" voice.codecPref.G729AB="3" voice.codecPref.IP_4000. G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/> And ulaw is set as preferred in the extension. as shown in sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown Thanks again, Andres ------------------------------
I guess the line 303096 is the more relevant, but I don't know where to start troubleshooting it.
Line 303095 is probably relevant, too. What codec is the phone configured to try first? It looks like the phone is trying to use something asterisk doesn't understand, or is not configured for. Maybe set the phone to ulaw instead. Also, what dtmfmode are you using? Can we look at your sip.conf from asterisk, and the config files for your Polycom phone? _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
