[asterisk-users] g729a compatibility

[Elliot Murdock]

Hello!

My Goodness! Thanks for the help!

I did set up a SIP account for that device to allow G729, but it
doesn't seem to be associating the device with that account.  So that
is where my confusion came from.

I'll make the changes and see what happens.

Regards,
Elliot

On Thu, Jul 2, 2009 at 7:08 PM, Kevin P. Fleming<kpfleming at digium.com> wrote:
> 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
>
> _______________________________________________
> -- 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
>