[asterisk-dev] SIP The Asterisk bug tracker ::pleasethinktwicebeforeopening a report!

[Wai Wu]

What I was saying is that this feature should'nt be done automatically
in *, but rather by a bridging app that knows how to handle negotiation
between the end points. 

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian Jones
Sent: Thursday, April 06, 2006 11:51 AM
To: Asterisk Developers Mailing List
Subject: RE: [asterisk-dev] SIP The Asterisk bug tracker
::pleasethinktwicebeforeopening a report!

Any other opinions on this?

> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian 
> Jones
> Sent: Wednesday, April 05, 2006 2:57 PM
> To: Asterisk Developers Mailing List
> Subject: RE: [asterisk-dev] The Asterisk bug tracker :: 
> pleasethinktwicebeforeopening a report!
> 
> Why not?  How do you handle faxing from an IAD when handing off to 
> another SIP provider such as Level3?  ulaw passthrough should work 
> under these circumstances. In most cases, I would want 729 for voice 
> calls, but this breaks it for faxes.
> 
> Brian
> 
> > -----Original Message-----
> > From: asterisk-dev-bounces at lists.digium.com
> > [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Wai Wu
> > Sent: Wednesday, April 05, 2006 2:46 PM
> > To: Asterisk Developers Mailing List
> > Subject: RE: [asterisk-dev] The Asterisk bug tracker :: 
> > please thinktwicebeforeopening a report!
> > 
> > I don't really think * should handle this. However, an
> option in the
> > dial command that allows you to specify a list of codec would be 
> > helpful.
> > 
> > -----Original Message-----
> > From: asterisk-dev-bounces at lists.digium.com
> > [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Brian 
> > Jones
> > Sent: Wednesday, April 05, 2006 2:46 PM
> > To: Asterisk Developers Mailing List
> > Subject: RE: [asterisk-dev] The Asterisk bug tracker :: 
> > please thinktwicebefore opening a report!
> > 
> > I have a feature request, but I don't want to put it in bug tracker 
> > unless someone here tells me it's OK.
> > 
> > I've already asked about this in #asterisk and on the
> forum, but never
> > received any help.  At least I don't believe this is a -users 
> > question.
> > 
> > I would like a way to do transparant codec negotiation.  
> For example,
> > I have one peer (peer A) that supports both ulaw and 729,
> and another
> > peer (peer B) that only supports ulaw.  As far as I can
> tell, there is
> > no way to force peer A to setup a call from peer B with
> ulaw instead
> > of 729, since that is the preferred codec for peer A.
> > 
> > What happens is this:
> > 
> > Call comes in from peer B and the SDP only offers ulaw.  
> > Asterisk invites peer A, and offers both 729 and ulaw.  It then 
> > reports a combined capability of ulaw, but then proceed to try to 
> > bridge the calls and transcode to 729.  This then fails, because I 
> > don't have 729 licensing on this particular server (and
> don't need it,
> > because I actually don't want to transcode). At this point,
> instead of
> > falling back to 711, the call fails.
> > 
> > Would a way to either fall back to other codecs or a way to force 
> > "transparent" codec negotiation be a legitimate feature request for 
> > the bug tracker?
> > 
> > Thanks,
> > Brian
> > 
> > ----------------------------
> > Brian Jones
> > Director, Technical Services
> > http://www.kanchalra.com
> > ----------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev