IMO it is a bridging problem. The evidence of this is: SIP -> Analog - no outgoing audio connection Analog -> SIP (actually Analog -> * -> SIP - everything ok.
Try putting an Answer() in front of Dial() in your dialplan (extensions.conf) and see if this goes away. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Adam Robins Sent: Wednesday, January 28, 2009 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropping incompatible voice frame 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
