Le dimanche 15 novembre 2009 à 23:45 +0100, Eric van der Vlist a écrit :
> Weirdly, they seem to be coming from the context I am using to define
> outgoing calls rather than the one for ingoing ones (like in asterisk
> 1.4), but I guess that's another issue!
Hmmm... I wonder where it can be docu
Leif,
Le dimanche 15 novembre 2009 à 22:44 +0100, Eric van der Vlist a écrit :
> Leif,
>
> Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :
>
> > I'm not sure you've provided enough of the trace here. It finds the peer,
> > but
> > rejects it with a 401 Unauthorized, which is
Leif,
Le dimanche 15 novembre 2009 à 16:18 -0500, Leif Madsen a écrit :
> I'm not sure you've provided enough of the trace here. It finds the peer, but
> rejects it with a 401 Unauthorized, which is not uncommon. And I don't see
> any
> authentication information in the first INVITE. This is w
Eric van der Vlist wrote:
> After a migration to asterisk 1.6, I don't receive sip incoming calls
> anymore.
>
> As fas as I understand the SIP debug traces, my server receives the
> request and reject it:
>
> ++
> <--- S
I can not help you much, but only confirm that SIP call from one of my provider
in Poland is not working.
Registration goes through OK but call does not go through.
Back to 1.4 version is the solution.
--
Joseph
On 11/15/09 19:05, Eric van der Vlist wrote:
>After a migration to asterisk 1.6, I d
After a migration to asterisk 1.6, I don't receive sip incoming calls
anymore.
As fas as I understand the SIP debug traces, my server receives the
request and reject it:
++
<--- SIP read from UDP:212.27.52.5:5060 --->
INV
