[Asterisk-Users] RE: SIP i.e. Is something broken?

[Dave Weis]

On Mon, 29 Sep 2003, WipeOut wrote:
> Dave Weis wrote:
> >On Mon, 29 Sep 2003, Brian Capouch wrote:
> >>Christopher J. Wolff wrote:
> >>>Is it safe to assume that a fresh CVS build will not have the SIP
> >>>translation problem described?
> >>>Just FYI: I had similar problems for a while, and then I completely 
> >>>scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). 
> >>>That solved the problem.
> >>I "checkout" rebuilt from CVS last night about 10:00 pm, and filed a bug 
> >>report at that time.
> >>It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my 
> >>Budgetone won't work with the "broken" code. . .
> >>It's as described in other mails--I can receive calls on the Budgetone 
> >>but when I make them the RTP part is broken and the calls cut off the 
> >>second they're set up.
> >I'm seeing the same thing with my budgetones and today's cvs. They are all 
> >on the same network and worked on previous versions. I can call 
> >voicemailmain and the console says that it is playing but I hear no sound. 
> >Then it hangs up automatically.
> So the issue looks like it is to do with the Bugetone phones and the 
> problem seems to have been introduced between Thurday last week and Sunday..
> The reason I say Thursday last week is becasue I checked out a fresh 
> copy of the CVS today with the -D "last Thursday" switch and its working 
> with the Bugetones..

That did it for me also. I did cvs co -D "last Thursday" asterisk. A rough 
examination of the diff between the two chan_sip.c's show that work was 
done on the codec negotiation. I think a bit more works needs done ;-)

-- 
Dave Weis             "I believe there are more instances of the abridgment
djweis at sjdjweis.com   of the freedom of the people by gradual and silent
                      encroachments of those in power than by violent 
                      and sudden usurpations."- James Madison