Summary:ASTERISK-03754: G726-16, G726-24, and G726-40 passthrough
Reporter:bmccrary (bmccrary)Labels:
Date Opened:2005-03-23 14:03:14.000-0600Date Closed:2011-06-07 14:05:21
Versions:Frequency of
Description:It would be really great if Asterisk supported SIP passthrough on G726-16, G726-24, and G726-40 codecs like is currently supported on G729 if a G729 codec is not installed.  Presently, I get the following message when trying to establish a G726-16 call:

Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port
Found description format G726-16
Found description format telephone-event
Capabilities: us - 0x114 (ulaw|g726|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Mar 23 15:02:06 NOTICE[29176]: chan_sip.c:3001 process_sdp: No compatible codecs!
Transmitting (no NAT) to
SIP/2.0 488 Not acceptable here
Comments:By: Russell Bryant (russell) 2005-03-23 14:07:46.000-0600

This is obviously not a bug.