[asterisk-users] Dropping incompatible voice frame

[Danny Nicholas]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 [2120 at 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.

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