[asterisk-dev] SIP The Asterisk bug tracker ::
pleasethinktwicebeforeopening a report!
Brian Jones
bjones at kancharla.com
Thu Apr 6 08:50:36 MST 2006
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
> > ----------------------------
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