Thanks, placing: Disallow=all Allow=ulaw
In the specific iaxy device context fixed it. I had always thought that allowing all possible valid codecs under the general context would work and the devices would sort it out upon handshake. Guess not. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Steven J. Douglas Sent: Wednesday, January 28, 2009 9:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropping incompatible voice frame Don't use g729 in the iax.conf for the IAXY device. It doesn't support it. Regards, Steve Adam Robins wrote: > I am using a Polycom SIP phone (ext 2042) to call an analog phone > connected via an IAXY (ext 2120). The analog phone rings, and when I > answer, I can hear the person speaking on the SIP phone, but they cannot > hear me. However, if I originate the call from the analog phone to the > SIP phone, it works just fine. > > In SIP.conf: > Disallow=all > Allow=g729 > Allow=ulaw > Canreinvite=no > > In IAX.conf: > Disallow=all > Allow=ulaw > Allow=g729 > Transfer=no > Codecpriority=host > > CLI shows: > > [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- > Executing [2...@international:1] Dial("SIP/2042-b7b0cc88", > "IAX2/2120|12|oWwtT") in new stack > [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- > Called 2120 > [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- Call > accepted by 192.168.2.61 (format ulaw) > [Jan 28 16:04:31] VERBOSE[21750] logger.c: [Jan 28 16:04:31] -- > Format for call is ulaw > [Jan 28 16:04:31] VERBOSE[3428] logger.c: [Jan 28 16:04:31] -- > IAX2/2120-3849 is ringing > [Jan 28 16:04:33] VERBOSE[3428] logger.c: [Jan 28 16:04:33] -- > IAX2/2120-3849 answered SIP/2042-b7b0cc88 > [Jan 28 16:04:33] NOTICE[3428] channel.c: Dropping incompatible voice > frame on IAX2/2120-3849 of format g729 since our native format has > changed to 0x4 (ulaw) > [Jan 28 16:04:41] VERBOSE[3428] logger.c: [Jan 28 16:04:41] -- > Hungup 'IAX2/2120-3849' > > This is Asterisk 1.4.22, but it also happened on 1.2.4. If I call an > IAX2/ulaw softphone from the SIP phone, it works fine. Could it be > something in the IAXY provisioning? > > Any ideas are appreciated. Thanks. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
