Daniel Pocock wrote:
On 01/04/13 22:06, Joshua Colp wrote:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words,
On 31/03/13 23:43, Joshua Colp wrote:
Daniel Pocock wrote:
I'm trying to call from DruCall to Asterisk and I get this error:
WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
103 104 111 0 8 107 106 105 13 126'
== Problem setting up ssl connection:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words, ignored or silently removed), as described here:
From a
On 01/04/13 22:06, Joshua Colp wrote:
Daniel Pocock wrote:
Thanks for the fast reply. I agree backporting full support for AVPF
would not be justified for many use cases (including my own). What I
was more curious about is whether the F can be tolerated (in other
words, ignored or
On 17/12/12 13:34, Joshua Colp wrote:
Barco You wrote:
Dear All,
Hola,
I use sipml5 to register two users from browser and the two clients
are successfully connected. But when I made a call from one of the
users, the other user doen'st have call notification and for a while the
Daniel Pocock wrote:
I'm trying to call from DruCall to Asterisk and I get this error:
WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
103 104 111 0 8 107 106 105 13 126'
== Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
I'm guessing my
Barco You wrote:
Dear All,
Hola,
I use sipml5 to register two users from browser and the two clients
are successfully connected. But when I made a call from one of the
users, the other user doen'st have call notification and for a while the
calling process ended. I check the
Dear All,
I use sipml5 to register two users from browser and the two clients are
successfully connected. But when I made a call from one of the users, the
other user doen'st have call notification and for a while the calling
process ended. I check the /var/log/asterisk/messages got the
