hello list,
i need your help about cdr ,i have installed the module cdr in my asterisk
1.4 .
for the inbound calls when i call my sip exten like below :
exten = 506,1,Dial(SIP/223, 10)
exten = 506,n,Dial(SIP/276, 10)
in CDR i have just one line with SIP /276 the last line but there is
no
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
hello list,
i need your help about cdr ,i have installed the module cdr in my
asterisk 1.4 .
for the inbound calls when i call my sip exten like below :
exten = 506,1,Dial(SIP/223, 10)
exten = 506,n,Dial(SIP/276, 10)
thanks i verify but i don't understanding if can someone give me an example
best regards
2013/5/9 Ishfaq Malik i...@pack-net.co.uk
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote:
hello list,
i need your help about cdr ,i have installed the module cdr in my
asterisk
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 can use
506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want
to do failover the check Dial
Hi,
Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus:
Am 07.05.2013 18:23, schrieb Sebastian Niehaus:
For some reason, t38modem tells hylafax the line is BUSY so there is no
fax send.
Well, I may add the log of t38modem (sorry for the ugly formatting)
Parts I consider
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 variables along
with a Newexten event. There may be a ton of extra variables that
someone may not want or need on the AMI. Better to have them ask
Hi Bilal, if you're looking for Asterisk, CRM and even Google Contacts
integration, do check out Aptus FonB (www.aptus.com).
I believe that's the exact solution you're looking for.
Br,
Maax
__
Hi;
I used vicidial for call
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 variables along
with a Newexten event. There may be a ton of extra variables that
someone may not want
Running Asterisk 10.12.2 on Debian/sparc
i'm doing all sip/rtp.
directmedia=yes
directrtpsetup=yes
I frequently see on the console:
WARNING[7832]: chan_sip.c:19134 show_chanstats_cb: Could not get RTP stats
What is this error trying to tell me ? 'sip show channelstats' shows
all 0s (save
Hello,
i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.
Thanks in advance.
-Motty
--
_
-- Bandwidth and
On Saturday, May 11, 2013 , the Asterisk community services listed below
may have intermittent availability due to routine maintenance being
performed. This maintenance will begin at approximately 12:00 AM CDT
(05:00 May 11 UTC)[1] and should last no longer than five hours.
The affected
Monitor what parts exactly?
Right this moment I'm in the process of installing Munin and the Asterisk
plugin to monitor channel usage, SIP connections, and the like. The Munin
server is running on a separate machine with just the node software on
Asterisk.
On Thu, May 9, 2013 at 12:23 PM,
Thanks for the suggestion Carlos,
do you have a HowTo? can you point me to one.
I unsuccessfully follow one found using google. I'm using CentOs 6.0
Thanks,
Motty
On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote:
Monitor what parts exactly?
Right this moment I'm
I'm using opennms and It's working fine.
On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:
Hello,
i'm looking for suggestions to monitor Asterisk Server? I installed Nagios
but no success, I do prefer not to install any web server on the server
running Asterisk.
http://opennms.org/wiki/Installation:Yum
On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote:
I'm using opennms and It's working fine.
On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote:
Hello,
i'm looking for suggestions to monitor Asterisk
It's not quick or simple, but there's decent documentation. I haven't been
saving the links I used, so I can't just give you specific places to look,
other than the best Asterisk plugin:
https://github.com/munin-monitoring/contrib/blob/master/plugins/asterisk/asterisk
TIP: Use chmod 755 on the
Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor Asterisk, not
the server Asterisk in running on.
thanks,
-Motty
On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Then you want a queue manager and reporting tool. Usually when people say
monitor Asterisk is has to do with the state of the system itself. You
should look at http://www.asternic.net and similar products. Munin will
tell you channels in use, but not the other stuff you want.
On Thu, May 9,
You can use queue-stats
http://www.asternic.org/stats/demo/
they has a free version
On Thu, May 9, 2013 at 4:12 PM, motty cruz motty.c...@gmail.com wrote:
Thanks for your help; I just want to monitor the queue, calls on hold
average time, incoming out going call, I only want to monitor
Howdy,
Looking to port numbers that we currently own in the US Virgin Islands
to *any* carrier that can do it in the states. XO says it is out of
their scope. Our normal DID carrier (IP Comms) apparently uses XO...
Looking for recommendations?
Thanks,
j
--
Queuemetrics works well for this also, and can be installed on a separate
machine/VM.
www.queuemetrics.com
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Thursday, May 09, 2013 3:13 PM
To: Asterisk Users
There is nagios plugin
check_asterisk_channels
Examples:
Check channels/calls, with no concern about limits.
check_asterisk_channels
Check channels/calls. Issue a warning if there are more than 10 active
channels, and a critical if there are more than 15 active channels.
thank you for your question because i research some topics about call
center with asterisk
can you please give a tutorial and the method that you use to implement
this call center with vividial .
2013/5/9 Maaz Bin Mahmood pointed@gmail.com
Hi Bilal, if you're looking for Asterisk, CRM and
My Google-Fu skills have failed me, I have not been able to find a solution
to the problem I am facing.
asterisk + from + asterisk + options + qualify != what I am looking for
--
When qualify is enabled on a trunk, the From line shows asterisk. See the
SIP message below.
I would like to keep
On 5/9/13 8:21 PM, Brian LaVallee wrote:
When qualify is enabled on a trunk, the From line shows asterisk. See the
SIP message below.
I had the same annoyance/issue. fixed it in
https://issues.asterisk.org/jira/browse/ASTERISK-17616
the patch was included in 1.8.9 rc1.
--
Jeremy Kister
On Thursday, May 09, 2013 8:23 PM, Jeremy Kister wrote:
On 5/9/13 8:21 PM, Brian LaVallee wrote:
When qualify is enabled on a trunk, the From line shows asterisk. See
the SIP message below.
I had the same annoyance/issue. fixed it in
https://issues.asterisk.org/jira/browse/ASTERISK-17616
Thanks Jeremy!
On 5/9/13 8:21 PM, Brian LaVallee wrote:
When qualify is enabled on a trunk, the From line shows asterisk. See the
SIP message below.
I had the same annoyance/issue. fixed it in
https://issues.asterisk.org/jira/browse/ASTERISK-17616
That's looks like the problem I was
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