Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a
Grandstream phone. Challenge here is I'd like to have Kamailio and
rtpengine to handle the bridging between different rtp profiles but
Asterisk
Hello
The answers to your questions are:
1, OS
CentOS release 5.5 (Final)
Trixbox installed at: Autogenerated by /usr/sbin/dahdi_genconf on Fri Nov 25
18:03:26 2011
2, Kernel
Linux 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:53:10 EST 2010 i686 i686 i386
GNU/Linux
3, Packages
Hi,
is there a way to put a conference participant in talk only mode (not
listening) using CLI or AMI like mute/unmute ?
MeetMe in Asterisk 1.8
Thanks for any hint.
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine with
Asterisk 1.8. I have managed to register and installed the Digium
modules. Sending and receiving through it have resulted in failure. The
output of fax show capabilities is:
Registered FAX Technology Modules:
Hello,
If the phone of my colleague rings, I can see this with BLF-lamps on my
Snom IP-phone. I would also like to see *_who_* is calling. I would like
to see the external number on my screen so I can choose whether to
pickup the call with BLF.
Therefore I have in sip.conf : notifycid = yes
Hello.
I've been trying to setup Free Fax for Asterisk on a Debian machine
with Asterisk 1.8. I have managed to register and installed the
Digium modules. Sending and receiving through it have resulted in
failure. The output of fax show capabilities is:
Registered FAX Technology
Hello,
Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent
Bleep (a private P2P SIP-based messaging application in early alpha)
http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/
I have personally been a fan of Asterisk and have been
On 11/08/14 16:46, Farid Fadaie wrote:
Hello,
Full disclosure: my name is Farid Fadaie and I'm in charge of
BitTorrent Bleep (a private P2P SIP-based messaging application in
early alpha)
http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/
I have
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come
through the same peer all the time, from the same carrier. However
intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful call (2nd). I
can't see any difference until we
On Monday 11 Aug 2014, Farid Fadaie wrote:
Hello,
Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent
Bleep (a private P2P SIP-based messaging application in early alpha)
http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-
communications/
I
Hello;
Just taking a quick glance at it, I think you have a syntax error in
your dial plan. Instead of ReceiveFax(${FAXDEST}/${tempfax}.tiff,f,d,
shouldn't it be ReceiveFax(${FAXDEST}/${tempfax}.tiff,fd) with no comma
between the f and d options?
Regards;
John
From:
On 8/11/14, 10:57 AM, David Wessell wrote:
I have an asterisk 1.8.x box that intermittently returns a 401. Calls
come through the same peer all the time, from the same carrier.
However intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 13.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users of Asterisk are encouraged to participate in the
Asterisk 13
On 11-08-14 11:09, Toney Mareo wrote:
Hello
The answers to your questions are:
1, OS
CentOS release 5.5 (Final)
That version is ancient and full of security holes. It is recommended to
at least update to CentOS 5.10 + updates. That's assuming there are
Trixbox kmod-dahdi-linux* RPMs for
On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
I'm trying to get calls working between websocket clients and sip clients.
For clients I have sip.js based clients on chrome, Zoipers and a Grandstream
phone. Challenge here is I'd like to have
Hello Asterisk,
Monday, August 11, 2014, 2:45:00 PM, you wrote:
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 13.0.0. This release is available for immediate download at
In my living dangerously mode, V13 beta 1 is up and running on my home PBX
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