thanks asghar for your help and support and thanks ishfaq
2013/5/9 Asghar Mohammad asghar...@gmail.com
hi,
asterisk insert cdr when call is hangup and last dial statment,
i dont understatnd why you are using 2 dial statment on same extenstion?
if you you want dial to both extensions you
Thanks! Matthew and Dan.
On Thu, May 9, 2013 at 10:18 PM, Matthew Jordan mjor...@digium.com wrote:
On 05/09/2013 08:16 AM, Dan Cropp wrote:
I believe you will have to monitor for the Newexten event, then send an
AMI Getvar command.
It doesn’t make sense to pass all the possible channel
We use Zabbix as monitoring tool and SNMP to get statistics and other
info from Asterisk.
for this you will have to make sure the snmp module for asterisk gets
compiled and the Asterisk MIB is used.
Regards,
Michel.
On 09-05-13 21:23, motty cruz wrote:
Hello,
i'm looking for suggestions to
Try with http://www.observium.org (Observium).
You can customize script to report into Observium's dashboard.
Regards,
On Fri, May 10, 2013 at 7:55 AM, Michel Verbraak mic...@verbraak.orgwrote:
We use Zabbix as monitoring tool and SNMP to get statistics and other
info from Asterisk.
for
Hi,
I want to configurate asterisknow 3.00 to send voicemail to email. I do not
know how to setup server to send email.Please help me
Regard
Boris Rouliane
Date: Fri, 10 May 2013 08:10:09 -0430
From: faston...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS
codec, which is part of the WebRTC standard as the default codec.
Thank you,
--
James Mortensen
Project Manager, VoiceCurve, Inc.
866-707-4590
james.morten...@voicecurve.com
--
On 13-05-10 02:45 PM, James Mortensen wrote:
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS
codec, which is part of the WebRTC standard as the default codec.
Doubt it.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
Ntegrated Solutions in Dallas, TX is still looking for voice guy. This
position is for US hire only, will not sponsor H1B work visa.
http://www.ntegrated.net/careers/
Thanks.
JR
--
JR Richardson
Engineering for the Masses
--
Hi folks,
What I trying to do here is exactly this:
http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html
My provider given me a Huawei modem which have 2 phone jacks on it, but instead
of using it I rather redirect my POTS number to my PBX. I ran into couple
hi,
you can try to change sip user agent and sdp session s , owner in sip
config same as your phone,s (modem).
asterisk by default send user agent = asterisk version , s= asterisk , o=
asterisk.
some providers are not happy if they see asterisk word :)
On Sat, May 11, 2013 at 12:27 AM, Sergej
Anyone here using Level 3 or ATT wholesale sip terminations services? I
would like to know on any minimums they would require? Also, an idea of how
competitive the rates are. I am not asking to disclose your custom rate
deck, just a what to expect. Finally, if you guys can PM me contact info
to
I've set up google voice to chat with me:
Forwards calls to:
me@gmail.com
and xmpp:
[general]
debug=no; Enable debugging (disabled by
default).
autoprune=yes ; Auto remove users from buddy
list. Depending on your
Sorry to chime in here, is it possible to change the Server: Asterisk
, s=Asterisk, and o= within sip.conf? What are the directives
exactly please?
Thanks in Advance,
Nick.
On 5/10/13, Asghar Mohammad asghar...@gmail.com wrote:
hi,
you can try to change sip user agent and sdp session s ,
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