Hi,
I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an
ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only
ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the
client asterisk with the server asterisk as IAX2 peer and want to connect
i would like to read information from a file (txt)
On Mon, Apr 28, 2014 at 9:29 PM, Rusty Newton rnew...@digium.com wrote:
On Thu, Apr 24, 2014 at 6:34 AM, binary dreamer
dreamer.bin...@gmail.com wrote:
hello everyone.
I am running plain asterisk and I am using asterisk's internal
Hi all,
I'd like to verify whether an Asterisk behaviour is expected or not, and
ask for advice for the best solution.
I have Asterisk 1.8.17.0 on debian wheezy, listening on UDP and TCP 5060,
and TLS 5061.
Asterisk is part of a dispatcher set in Kamailio (4.1.3), and is marked as
AP (Active
On Tue, Apr 29, 2014 at 1:06 AM, Xengis Khan xengisk...@gmail.com wrote:
Hi,
I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an
ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only
ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure
This may be a more phone-specific question, but figured I'd ask to see if
someone has experience with this. I have a SIP phone (Polycom) configured
with two lines registered to two different Asterisk servers. I have
successfully configured SIP subscriptions to watch a different phone
registered
Hello,
Is there an implementation for the RFC 4662 for asterisk 10? I found a
patch for asterisk 1.8 but nothing for asterisk 10.12.
The RFC: This document presents an extension to the Session Initiation
Protocol (SIP)-Specific Event Notification mechanism for subscribing
to a homogeneous
I just finished migrating our web interface from Mysql to SQlite3
and everything seems to be working fine. I just have one detail. The
following keeps appearing on my logs:
[Apr 29 13:09:32] WARNING[30494]: res_config_sqlite3.c:520
realtime_sqlite3_execute_handle: Could not execute
Hello,
If anyone has successfully compiled asterisk with:
app_rtsp
codec_amr
mp4_play
mp4_save
app_transcode
h324m_call
Please share the versions of OS software, and libraries used.
Lets make this thread useful so that all tried and tested video resources
of asterisk can be found in one place
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en')
-- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge
5edb1920-3774-4ba3-8c4d-23e8fd04519c
--
On Tue, Apr 29, 2014 at 5:10 PM, Richard Kenner ken...@gnat.com wrote:
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language
'en')
-- Channel
On 29-04-14 20:41, [Digital^Dude] ® wrote:
Hello,
If anyone has successfully compiled asterisk with:
app_rtsp
codec_amr
mp4_play
mp4_save
app_transcode
h324m_call
Please share the versions of OS software, and libraries used.
Lets make this thread useful so that all tried and tested video
The announcer channel joins/leaves the conference as it has sounds
to play. If the channel still hangs around after the conference is
destroyed then there is a problem.
There's a problem. ;-)
But thanks for pointing to how that's supposed to be handled.
--
If the channel still hangs around after the conference is destroyed
then there is a problem.
Am I missing something obvious: I'm looking in the confbridge_exec
function. I see a conference = NULL line, but no attempt to free
that structure, which is what I understand will destroy the playback
Hello,
I am trying to diagnose an intermittent error when a call comes in over our
PRI lines.
The problem appears random, however I have feeling it has something to do
with the call volume, as the frequency increases with more calls on the
system.
I am not an expert when it comes to
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