[Asterisk-Dev] Codec not negotiating

[Michael Giagnocavo]

In my patch, I think I have a special value that CODEC_OVERRIDE can be:
NATIVE. If you set that, then when an IAX call goes out, it'll take the
current native format and request that only. I forget if there's a SIP
implementation. Have a look at the override patch :)

-Michael

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Clay Reiche
Sent: Monday, April 04, 2005 1:46 PM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Codec not negotiating

I set up the multiple peers and that is a good enough fix for now. Thank
you!
Now, how would I do this if my SIP device was smart enough to determine the
call type on the fly? Ie...voice(g729) or fax(ulaw). The codec being sent by
my SIP device(peer) would change depending on the call type. This multiple
peer solution would no longer work.

Clay

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jerris, Michael
MI
Sent: Monday, April 04, 2005 3:04 PM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Codec not negotiating

That bug does create an override codec (it is named poorly in the bug
description).  Multiple peers would certainly work. 

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Michael
Giagnocavo
Sent: Monday, April 04, 2005 2:56 PM
To: 'Asterisk Developers Mailing List'
Subject: RE: [Asterisk-Dev] Codec not negotiating

I'm not even sure if there will ever be a way to restrict the codecs sent.
Some people seem to think just sending a preferred codec is a good solution
when in reality to force a preferred codec, you must only send that codec.

The easy solution here is to create two peer entries, one for ULAW, and one
for G729 and then dial one or another.

-Michael

________________________________________
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jerris, Michael
MI
Sent: Monday, April 04, 2005 12:23 PM
To: Asterisk Developers Mailing List
Subject: RE: [Asterisk-Dev] Codec not negotiating

http://bugs.digium.com/bug_view_page.php?bug_id=0003346 should address this
issue, but there is not yet a patch with the implementation that was decided
upon yet.

________________________________________
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Clay Reiche
Sent: Monday, April 04, 2005 1:57 PM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] Codec not negotiating ok... I've trying to fix this
for days... I got very little response from the Users list. I have a sip
device that registers with my *. The sip device is ONLY set up to use ulaw.
My asterisk server sends ALL PSTN calls to a Sonus gateway/softswitch. When
I
place a PSTN call, the sip device sends the INVITE with SDP and the ONLY
codec option is ulaw. Asterisk then turns around and sends an INVITE with
SDP
to the Sonus gateway with ulaw as the first option and g729 as a second
option. The Sonus sees the TWO options and ALWAYS chooses g729. The codec
negotiation fails and the call never completes.
 
I understand that the TWO options are sent because I have no peer set up for
the Sonus in my sip.conf and it defaults to the [general] codec settings
which are ulaw and g729. However, MOST of my calls to the Sonus ARE using
g729, only a few need to use ulaw. (for faxing) So I can't restrict the
Sonus
peer to only ulaw...
 
Here is my question:(finally...sorry:))
Can I force asterisk to send ONLY my prefered codec?(the first one in the
INVITE) or is this only fixed by pleading with the people who run the Sonus
sofswitch to stop ignoring my preferred codec? or is there some other
solution? Any suggestions would be very appreciated!
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