On Friday 17 May 2013, cjwstudios wrote:
A friend asked me for help to auto-dial and play a prerecorded message for
a political campaign. I've briefly googled auto dialer scripts but haven't
seen one that really stands out. Are there any free or cheap auto dial
solutions that you nice folks
On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:
Hi,
I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is
generating are failing. I am trying to run Sipp on the same machine as
Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
Do you have a peer
On 22/5/13 10:54 am, A J Stiles wrote:
You do know that sort of thing is against the law -- or at least requires a
permit from the authorities -- in most civilised countries, right?
And it's worth adding that even if it is legal in your country, you're
almost guaranteed to offend/annoy your
Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Thank you for your help I finally solved this issue. Is it possible that my
setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core
using 3.5 GHz, and 1Gb of RAM?
- Forwarded Message -
From: Marie Fischer ma...@vtl.ee
To: Asterisk Users Mailing List -
I have a question here.
How can we test the quality of voice upon increasing the call load?
Can we try passing a voice file using sipp and record the same in dial plan
record application ? Is this reliable enough to simulate near real world
scenario?
Mitul
On Wednesday, May 22, 2013, Tommy
You may have noticed (or maybe not) that there have been several
maintenance notifications for the asterisk.org community services this
month. We are working hard to keep up the services running smoothly,
and those notices are sent whenever we think our maintenance may
interfere with the operation
From the little experience I have I do not think that that is a good way of
testing the quality of voice. SIP only initiates and eventually terminates the
call, once that the call is connected, SIP and therefore Asterisk are no
longer involved. Once the call is connected it is assigned to a
I believe there are options for rtp / audio..
Look at pcap play and rtp echo...
Transcoding would be another beast - if you are allowing it
Sent from my iPhone 5
On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote:
From the little experience I have I do not think that that
Hi guys,
Any idea why I am getting this error when someone tries to send me a T38
Fax?
--
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On Wed, 22 May 2013, A J Stiles wrote:
If the call file is definitely smaller than one block (the size of which
depends on your file system), it should be OK to write in situ.
Otherwise, write it under /tmp or somewhere and then use the system
command mv to move it to
On 13-05-22 10:02 AM, Tommy Cooper wrote:
From the little experience I have I do not think that that is a good way of
testing the quality of voice. SIP only initiates and eventually terminates the
call, once that the call is connected, SIP and therefore Asterisk are no longer
involved. Once
El 22/05/13 12:25, Paul Belanger escribió:
On 13-05-22 10:02 AM, Tommy Cooper wrote:
From the little experience I have I do not think that that is a good
way of testing the quality of voice. SIP only initiates and
eventually terminates the call, once that the call is connected, SIP
and
asterisk users wrote:
Registration trace
(note that extension 88 is the voicemail extension, which the phone registers
to also for MWI)
-- http://pastebin.com/c3H700wa
There are no REGISTER requests in that trace. All I see are SUBSCRIBE, NOTIFY,
OPTIONS, and INVITE dialogs.
Call trace:
Jim,
Cron and Logrotate already installed in my machine and already configured
as the steps you enlisted. But still logrotate is not running.
Date: Tue, 21 May 2013 12:28:31 -0700
From: Jim Lucas li...@cmsws.com
Subject: Re: [asterisk-users] Asterisk Log rotate not working
To: Asterisk Users
On 22.05.2013, at 16:18, Tommy Cooper tomcoope...@yahoo.com wrote:
Thank you for your help I finally solved this issue. Is it possible that my
setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core
using 3.5 GHz, and 1Gb of RAM?
Easily, as long as you have no media :)
We have a scenario where we wish to present a toll-free caller id, yet have
our calls rated based on our billing-telephone-number. Is it possible to
present a number in the sip header for billing and another number in the
header for jurisdicional call rating?
Whereas today, all of our calls are
On Wed, May 22, 2013 at 02:54:46PM -0400, Ahmed Munir wrote:
Jim,
Cron and Logrotate already installed in my machine and already configured
as the steps you enlisted. But still logrotate is not running.
How can you tell that the logrotate cron job was run?
At what time it was configured to
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