Am 22.07.2013 um 17:26 schrieb Tom Chadwin :
> Hi Michael
>
> I'm not using the r option in the dial command (we use TtHh). Also, calls
> via our berofix work correctly - it is only internal SIP calls (between
> Snom300/320s) which exhibit the behaviour.
>
> We were using Asterisk 1.8 before, w
Hi Michael
I'm not using the r option in the dial command (we use TtHh). Also, calls
via our berofix work correctly - it is only internal SIP calls (between
Snom300/320s) which exhibit the behaviour.
We were using Asterisk 1.8 before, whichever version was packaged in with
the immediately previou
Tom,
I don't know if my recent experience has anything to do with yours.
However, I saw a coincidental change in Asterisk behavior between 1.8.21
and 1.8.22 regarding session timers. Perhaps this is related.
Over the weekend, I upgraded from Asterisk 1.8.21 to Asterisk 1.8.22.
Following th
Tom,
Are you setting "directmedia=no" in your sip.conf for the local extensions ?
This supersedes the old "canreinvite=no" in Asterisk 1.4 .
Lonnie
On Jul 22, 2013, at 10:26 AM, Tom Chadwin wrote:
> Hi Michael
>
> I'm not using the r option in the dial command (we use TtHh). Also, calls
>
Hello all
We upgraded recently to the most recent 1.8 Astlinux, and we have a problem
we've not encountered before. Intermittently on internal SIP-SIP calls only,
there is no audio for a varying number of seconds (between 2 and 10) -
caller hears no ringing tone, and neither party can hear each ot
Am 22.07.2013 um 16:45 schrieb "Tom Chadwin" :
> Hello all
>
> We upgraded recently to the most recent 1.8 Astlinux, and we have a problem
> we've not encountered before. Intermittently on internal SIP-SIP calls only,
> there is no audio for a varying number of seconds (between 2 and 10) -
> cal
