For WebRTC, I recommend you to use Asterisk 13+.
Have a nice day.
Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introdu
The solution that fixed our problem was to Edit the
sip_general_additional.conf file by adding the line "session-timers=refuse"
Thank you to each one who gave suggestions.
Keith
Keith Heppner
Rio Grande Bible Institute
4300 S Business Highway 281
Edinburg, TX 78539-9650
Office 956-380-8171
Cell
Hello
I'm trying for several days now to get ICE support for my Asterisk 11.23
on CentOS 6.
My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230
--> softphone Zoiper
(problem : no audio)
Reverse does not work either.
(problem : failed get local SDP)
I followed this guid
On Tue, Aug 2, 2016 at 11:42 AM, nik600 wrote:
> Dear all
>
> i'm trying to access to the input audio raw stream with a very basic EAGI
> script:
>
>
> #!/bin/sh
> echo "EXEC Queue 2001"
> cat /dev/fd/3 > /tmp/pippo
>
> This is my dialplan:
>
> exten => 001,NoOp(test)
> exten => 001,n,Answer
> ex
On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote:
> Hi All,
>
> We have asterisk 11.23 running sip to vitelity and from there IAX trunks
> split off to where they need to go. We are having a problem getting
> chan_sip to quit ignoring re-invites from Vitelity. Our side ends up
> sending a rei
El 08/08/16 a las 21:34, Eric Wieling escribió:
How Set handles quotes can be changed with the 'app_set' setting in the
[compat] section of /etc/asterisk/asterisk.conf. See also:
https://wiki.asterisk.org/wiki/display/AST/Application_Set Perhaps you have the
value left over from an old Aster
On Sat, Aug 6, 2016 at 11:13 AM, Chirag Desai wrote:
> All,
>
> I upgraded to asterisk 13.10. I have minimal load on the box. 20-30 calls a
> day.
>
> Right now, there are no calls on the box at all.
>
> top shows me this:
>
> PR 20
>
> NI 0
>
> VIRT 1570540
>
> RES 84620
>
> SHR 26296
>
> S S
>
>
Jacek,
This might be a bug or configuration issue, but you need to understand the
SIP Session Timers. With Session Timers you can control the round trip time
and Call Setup time of SIP Requests.
In case of GSM Network with high delay you need to set the T1 timer a
higher value like 1000ms (500 ms d
Hi,
We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
stumbled on a behaviour difference I don't like.
With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
disconnected) Asterisk would detec