Please try our billing which has easier managing interface and works ok with
H323: http://www.voip-info.org/wiki/view/MOR
FREE version is available over this link:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing
Hi All;
My Asterisk version is: 1.4.19.1
I am not able to pickup a call ringing at SIP Phone (exten 802), the call was
transferred from Zap FXO channel and I am trying to pickup it from SIP Phone
(exten 800), but it fails with the below error:
[Feb 2 21:14:25] NOTICE[2703]:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
ibpthread-2.7.so[7f3b8e903000+16000]
Anything what can be done to find out the reason?
My asterisk 1.4.23
Tzafrir Cohen wrote:
On Fri, Feb 05, 2010 at 01:55:03PM -0500, sean darcy wrote:
sean darcy wrote:
Using 1.6.2.1 with a TDM400, attached to internal analog phones and
PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for
something stupid. The call itself works, but the DTMF
You want to set it like this on Asterisk:
tos_sip=cs3
tos_audio=ef
tos_video=cs4
And in Polycom config:
qos.ip.rtp.dscp=EF
qos.ip.callControl.dscp=24
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54
Nikhil Nair wrote:
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Thanks for posting this. And for persistently
Does your billing work with gnugk? Do u have a documentation on how it can be
used with gnugk?
Does the free version work with the gnugk?
Regards
Bilal
Please try our billing which has easier managing interface
and works ok with
H323:
I have the same problem. I have asterisk on the public internet and
other ips on the private lan. When the internet goes down my private
asterisk network is compromised. My thought is that it has something to
do with the ports/ips on which asterisk is trying to communicate. It
may be a
Hello my friends,
I'm having a problem like this post...the difference is that my asterisk
goes down and i have to reboot my server in order to make it up again...
following you will see some errors that i can see in the Asterisk
/var/log/messages qhen asterisk goes down:
[Feb 5 10:32:45]
Thanks for the replay...
I will check the memory tommorow with memtest86 from this site:
http://www.linux.org/apps/AppId_7360.html
I will let you know...but again, for me, the problem is in the network, may
be some problem with the DHCP or DNS server from the client...what do you
think?
--
On Sun, 7 Feb 2010, Danny Dias wrote:
I will check the memory tommorow with memtest86 from this site:
http://www.linux.org/apps/AppId_7360.html
memtest is on Knoppix, CentOS, and probably a bunch of other bootable
CD/DVDs you should have laying around.
--
Thanks in advance,
I'd vote for DNS problem myself. Do you have a local dns sevre that forwards
unknow requests?
--
Josiah Bryan
Productive Concepts, Inc.
jbr...@productiveconcepts.com
Cell: 765-215-0511
Desk: 765-215-6009 x224
-Original Message-
From: Danny Dias ing.diasda...@gmail.com
Date: Sun, 7 Feb
Sean Brady wrote:
Have you tried removing the /n option from the local channel? Just a
thought, but it's probably worth a try. You could also try calculating the
billsec in the dialplan and write it to the CDR with the adaptive CDR feature
in 1.6.2.
Not sure if this is helpful but it
I am trying to understand .call files.
The logs seems to indicate issues with the audio file that I am trying
to have played when the call is connected.
Any thoughts? Some sample files and logs to console are shown.
zipp-code.call
Channel: SIP/callwithus/12023519259
Application: Playback
Your sound file needs to be in the asterisk sounds directory.
Another thing is that you may not have to put the file extension in the name
if the file is in the proper place as well.
Try that and see what happens.
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi Tommy
Thank you
works like magic. thank you. I love this list. when you get stumped
you can always (almost!) count on some great input!
regards,
tom
On Sun, Feb 7, 2010 at 7:32 PM, Tom Moore tommym2...@gmail.com wrote:
Your sound file needs to be in the asterisk sounds directory.
Another
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas
Perron
I am trying to understand .call files.
The logs seems to indicate issues with the audio file that I am trying
to have played when the call is connected. Any thoughts? Some sample
files and
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
server without having to copy them back and forth to the Asterisk
Hi All,
I am in the process of migrating from 1.4.20 to 1.6.2.x and have
stumbled across a number of differences between the 2 versions that
are forcing me to use local channels in my dialplan (mainly centered
around the different behavior of CDR fields in the 2 versions) .
Previously, I
Leo Burd wrote:
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
server without having to copy them back and
Thanks Josiah Bryan,
I do not have any dns server running on my asterisk server, we do have an
external DNS server working in the data center; the IP of this dns server is
10.4.1.5...
Following you will see my main configuration:
/etc/resolv.conf:
domain localdomain
search localdomain
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
At 09:51 2/7/2010, Vinícius Fontes wrote:
You want to set it like this on Asterisk:
tos_sip=cs3
tos_audio=ef
tos_video=cs4
Why cs4 instead of af41?
And in Polycom config:
qos.ip.rtp.dscp=EF
qos.ip.callControl.dscp=24
Thanks, Vinícius, but this is for Asterisk v1.4,
yes?
The
Not able to compile asterisk,zaptel,libpri in /usr/src--
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aster...@opensourcesolution.in wrote:
Not able to compile asterisk,zaptel,libpri in /usr/src
Have you tried to run make?
Without any information on what you're tried and what error you receive,
I can almost guarantee you will not receive any help on this forum.
Regards,
--
Trevor Peirce
Please see http://www.officeforlawyers.com/howask.htm
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com http://www.rolandschorr.com/
b...@rolandschorr.com mailto:b...@rolandschorr.com
From:
Dear All,
I am using asterisk 1.4.21.2. I have used Originate manager application
to to call the two persons. I have called AGI application to call another
person. There I have used GET FULL VARIABLE AGI command to get the value. I
am able to call another person form AGI. But when one end cut
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
ibpthread-2.7.so[7f3b8e903000+16000]
Anything what can be done to
On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson o...@edvina.net wrote:
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in
On Sun, Feb 07, 2010 at 07:47:35PM -1000, Ben Schorr wrote:
Please see http://www.officeforlawyers.com/howask.htm
You probably meant:
http://www.catb.org/~esr/faqs/smart-questions.html
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
ibpthread-2.7.so[7f3b8e903000+16000]
Anything what can be done to
On Mon, Feb 08, 2010 at 02:37:18AM -0500, Steve Totaro wrote:
Just start it with safe_asterisk.
http://linux.die.net/man/8/safe_asterisk
And I take it that the name is intentional.
Unless my info is out of date, it will kill two birds with one stone.
You're in a lethal mood today :-)
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