Hi
Allo also offer gateways - similar to Fonebridge. Have you
tried/evaluated that? It will help us to know those results.
What is CPU RAM utilization on this server?
What kind of work load you run? Does it involve transcoding (codecs
used)? Are these calls passed on to SIP client or these
i do have asterisk 1.8 (no gui, no distro based) and i would like to enable
some features:-call forward (conditional, unconditional,...)-DND-call
waiting-attended transfer-follow me
all the features i would like to enable/disable them through digit codes such
#45# and *45.all these fetures
Hi all,
I want to make initial VoIP authentication process from asterisk server to
be based on EAP-SIM authentication of Freeradius server (so it will be not
necessary to insert account datas in the asterisk database). Is there any
way of doing that from Freeradius and Asterisk? Or at least, is
Hello Folks,
I looking to migrate a pbx from one server to another. The original server has
this ISDN card:
00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller
[HFC-4S] (rev 01)
The new server:
00:00.0 I2O: Digital Equipment Corporation StrongARM DC21285 (rev 04) AVM
Hi Toney,
Comments inline.
On 07-08-14 12:10, Toney Mareo wrote:
Hello Folks,
I looking to migrate a pbx from one server to another. The original server has
this ISDN card:
00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller
[HFC-4S] (rev 01)
The new server:
00:00.0
To enable transfers using in-call DTMF sequences, you'll need to use the t
and/or T options in the Dial() command that initiates the call. For
details see:
https://wiki.asterisk.org/wiki/display/AST/Application_Dial
On Thu, Aug 7, 2014 at 2:29 AM, Aristeidis Tsitras tsit...@hotmail.com
This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
# asterisk -x core show channels
Channel Location State Application(Data)
SIP/thinktel-000 (None) Up AppDial((Outgoing
On Wednesday 06 Aug 2014, I wrote:
I'm trying -- unsuccessfully! -- to configure an inbound trunk with
Simwood, and I was hoping someone on this list might have managed to do
this.
I have configured some numbers to route to a SIP endpoint
%e164@customer's server
and convinced the
Your call is up on VoiceMail you should check dialstatus before sending
user to VoiceMail.
On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain da...@vex.net wrote:
This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
#
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
so
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
is incorrect now? That page says:
Unless there is
On Thu, 7 Aug 2014, A J Stiles wrote:
. And my mistake was in sip.conf. The configuration stanza I had named
simwood_in_slough should, of course, have been named after the number I had
programmed in at the other end of the trunk .
*hangs head in shame*
It's OK. We're all a little
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
I removed the voicemail command from the dialplan and it was exactly
the same behaviour.
--
D'Arcy J.M. Cain
System
On 8/7/14, 12:14 PM, D'Arcy J.M. Cain wrote:
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
I removed the voicemail command from the dialplan and it was exactly
the same
I am using a cyberdata sip paging adapter and with the
Dial(MulticastRTP/basic/IP:port) and with
tshark I see the RTP data, my device looks like its accepting the data
and I hear a click for my relay on my device so it would seem its accepting
the call,
however - I hear no audio...
Asterisk
Hi,
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user
You can use sip proxy servers on top of asterisk server to have a
authentication from freeradius, at this point I don’t think asterisk supports
what you are looking for.
Try this
http://www.opensips.org/Documentation/Tutorials-Radius
From: asterisk-users-boun...@lists.digium.com
Uncommenting features.conf is not sufficient, You got to have some logic on
the dialplan to support what you are asking for. If I were you, I would
probably use some dial plan logic with asterisk internal DB .
From: asterisk-users-boun...@lists.digium.com
may i have an example of what you are describing?
On 7/8/2014 23:13, Shishir Pokharel wrote:
Uncommenting features.conf is not sufficient, You got to have some
logic on the dialplan to support what you are asking for. If I were
you, I would probably use some dial plan logic with asterisk
http://www.voip-info.org/wiki/view/PBX+Do+Not+Disturb
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of aris tsitras
Sent: Thursday, August 07, 2014 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi Guys..
I am making an anoucement machine that is not allowed to answer the call
due to a billing issue.
I found that Playback with noanwser is usefull in this case.
$AGI-exec('Playback',$message,noanswer)}
But when i request some values to the user with get_data, i think there is
an answer
What you may want to check out is the PlayTones and Ringing applications in
your dial plan. Asterisk will answer the call, but your users won't know that
because all they hear is the call still ringing. After a certain amount of time
passes, you can send them directly to voicemail, hangup,
Hi John.
I am making an inteligent annoucement resouce for a big ericsson switch. Is
just an ivr with agi applications.
The tricky thing try to make asterisk not to send answer. The perl
application with agi commands must be executed with out answering.
Something like
exten = 6009,1,Progress()
Generally the only thing you are allowed to do before answer is send audio.
You can’t receive audio and can’t receive DTMF. I assume it is to prevent
people from doing exactly what you are trying to do --- trying to have two way
communications without paying for the call.
From:
I am seeing this in my log file
:[Aug 7 21:35:24] ERROR[19582] acl.c: Cannot create socket
[Aug 7 21:35:24] WARNING[19582][C-0283] res_rtp_asterisk.c: Unable to
allocate RTP socket: Too many open files
[Aug 7 21:35:24] NOTICE[19582][C-0283] chan_sip.c: Failed to
authenticate device
On Thu, 7 Aug 2014 22:00:47 -0400
Jerry Geis ge...@pagestation.com wrote:
:[Aug 7 21:35:24] ERROR[19582] acl.c: Cannot create socket
[Aug 7 21:35:24] WARNING[19582][C-0283] res_rtp_asterisk.c:
Unable to allocate RTP socket: Too many open files
...
I am running asterisk 11.11.0
Shot in
On Thu, 7 Aug 2014 10:12:02 -0400
D'Arcy J.M. Cain da...@vex.net wrote:
This just started after upgrading to 11.11.0. After a call is
completed (both ends hang up) the call still shows as active.
New data point - I just reverted to 11.10.2 without a single change to
the configuration and the
Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain:
New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.
Hmm. Could this have to do with session-timers (sip.conf)?
I remember when I went from 1.4 to 10.7 I had to manually mess with
Hi,
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user
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