[asterisk-users] g729a compatibility

[Kevin P. Fleming]

Elliot Murdock wrote:

> [Jul  2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) ---
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 :
> 5060 (no NAT)
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as
> basis request - 6998640000475636237-1246542986-18105
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Found no matching peer or
> user for '216.48.184.50:5060'
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 18
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 98
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 101
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 13
> [Jul  2 16:56:26] VERBOSE[13420] logger.c: Capabilities: us - 0x8000e
> (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing)
> , combined - 0x0 (nothing)

And there it is... you have not allowed G.729 to be used by that peer in
sip.conf. In addition, no peer or user in sip.conf was found to match
the request, so unless you can correct that situation, you'll have to
modify the allow/disallow options in the general section of sip.conf,
since this call is being handled as an 'anonymous' peer.

Asterisk properly parsed the SDP and understands that the peer supports
G.729. None of the concerns about SDP parsing or RFC compliance, as it
turns out, were even relevant to this problem :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org