Too early... call droped after 11 seconds now... different log.
[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: SIP TIMER: Rescheduling
retransmission #13781 (6) SIP/2.0 - 1
[Mar 17 22:19:17] DEBUG[2783] chan_sip.c: ** SIP timers: Rescheduling
retransmission 7 to 4000 ms (t1 100 ms (Retrans id #1378
Hello Gentleman,
I guess the problem was the codec.
I have allowed only g711u for testing purposes and the incoming call endured
for 1 minute, until the caller hanged.
Thanks a lot for the support but there are tons of questions yet to be
answered!
Thanks
On Wed, Mar 17, 2010 at 2:12 PM, F
On Mar 17, 2010, at 1:05 PM, Bruno Camargo wrote:
> Hi Giorgio,
>
> So it means that Asterisk has no native support for g729 ?
>
> Thanks
>
> --
> BrCaBadT
> --
Depends on your definition of support. It supports passthrough... but if you're
using it locally on a bridge on transcoding, you'l
Hi Giorgio,
So it means that Asterisk has no native support for g729 ?
Thanks
On Wed, Mar 17, 2010 at 7:04 AM, Giorgio Incantalupo <
gincantal...@fgasoftware.com> wrote:
> Hi Bruno,
>
> I remember one of our customer had a similar problem with tellfree in
> Brazil. Their IT technician told me i
Hi Bruno,
I remember one of our customer had a similar problem with tellfree in
Brazil. Their IT technician told me it was due to a g729 codec
problem...they installed it and the problem disappeared. I never
checked, I could only trust their man.
Maybe it can help.
Giorgio
P.S.: let me know i
Hello Gentleman,
I'm new to asterisk, this is my first instalation, first post... so I'd like
to apologize if this question has already been made. I googled but I
couldn't find nothing similar.
Here's the thing.
I'm migrating from ATA to Asterisk one of my client's office and I have a
very simpl
