Hello list,
where can I post the output of the trace taken from a file :
/tmp/core.sip.pbx.tld-2012-01-17T11:09:13+0100
I want someone to tell me what went wrong.
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation
Hi,
Now I'm using Asterisk version 1.6.22. I use two server cluster with
load balancing mode.
My problem is: If one phone from a pickup group is moved to another
server, he can not take over the group call and someone call he the
other pickup group members can not take over the his call.
Slowly
Hello list,
can I conclude that it is better to use sub's in stead of macro's ?
I read the following in an Asterisk-book :
GoSub() works in a different manner from Macro(), though, in that it
doesn't have the stack space requirements, so
it nests effectively. Essentially, GoSub() acts like
Well... It worked! I would have wrote back sooner but I got swamped
with trying to move a business department to the new facility.
I think my problem was that I fat fingered (actually, overlooked) the
trunk group. The d channels should have been 48 and 72 because I was
using span 2 and 3. I kept
On 16/01/12 07:59, Roi Stork wrote:
I also asked my provider to test call me using their Cisco as5300
system and g729 codec and compared it with ulaw. The difference is
unnoticable.
^^ this doesn't make any sense, the difference *should* be very much
noticeable. g729 is a lower quality
On 01/17/2012 05:16 AM, Louis Carreiro wrote:
Well... It worked! I would have wrote back sooner but I got swamped
with trying to move a business department to the new facility.
Great! Glad to hear it is working for you :)
I think my problem was that I fat fingered (actually, overlooked)
Thanks for your hint, but unfortunately this does not result in the
behaviour I am looking for. When I start MusicOnHold Asterisk
streams the OnHold music itself, even if I specifiy an invalid MoH
class or one without files.
What I was looking for is a way to send a re-INVITE to its upstream
SIP
On 01/16/2012 03:59 PM, Roi Stork wrote:
Hi,
We noticed a very sharp drop in voice quality when using digium g729a
codec. The problem seems to happen if the A channel (caller's channel)
is a landline/mobile number contacted using the same outgoing provider
(as a local channel). It sounds like
Jonas
From what I understand they are trying to phase out Macros. We are slowly
removing them from our dialplans as time allows for testing.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
I have written a program that monitors asterisk to make sure my peers and
channels are all in good order. The program calls asterisk once a min and
then parses the output. The program works fine when launched from the
command line. I then wrote a script to launch the program with the hope of
You want your program to live in /usr/local/bin. /etc/init.d is where the
bash scripts that run programs that live elsewhere are housed. It is not a
good practice to put executeables there. For example, /etc/init.d/asterisk
runs /usr/sbin/safe-asterisk. The scenario I typically use is that
On Tuesday 17 January 2012, Bryant Zimmerman wrote:
I have written a program that monitors asterisk to make sure my peers and
channels are all in good order. The program calls asterisk once a min and
then parses the output. The program works fine when launched from the
command line. I then
What version of Asterisk are you trying to implement this in?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, January 17, 2012 1:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Asterisk 1.4.27 using .
Best Regards,
Mahesh Katta
On Tue, Jan 17, 2012 at 8:01 PM, Danny Nicholas da...@debsinc.com wrote:
What version of Asterisk are you trying to implement this in?
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
On 12-01-17 05:16 AM, Jonas Kellens wrote:
Hello list,
where can I post the output of the trace taken from a file :
/tmp/core.sip.pbx.tld-2012-01-17T11:09:13+0100
I want someone to tell me what went wrong.
Here is usually fine.
--
Paul Belanger
Digium, Inc. | Software Developer
twitter:
Hi All,
I am writing a billing engine in AGI. My scenario is :
One customer can have simultaneous calls and I need to hang up one
customer's all call when balance reaches 0
If I set limit for each call using 'L' in dial command, lets say 5 minutes
in accordance with remaining credit and connect
Hi
I have an application. It connects to Asterisk via AMI, and when user
decides it begins asynchronous origination to some device. But very
often user decides to break origination and make another call. How can
I achieve that? As much as I see, Asterisk doesn't returns any ID of
dial process and
I've got some users reporting an odd problem.
Once in a while, their Polycom phones ring and they are unable to answer
them... any of them.
When they pick up the handset, all of the phones continue to ring. Same thing
happens if they grab a different phone. They aren't reporting any
On 17-01-12 18:23, Mike Diehl wrote:
I've got some users reporting an odd problem.
Once in a while, their Polycom phones ring and they are unable to answer
them... any of them.
When they pick up the handset, all of the phones continue to ring. Same thing
happens if they grab a different
Can they answer the call by pressing the line key when simply picking up the
handset does not answer the call? If so, then the users are not properly
seating the handset in the cradle.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I use the latest spandsp source from the freeswitch git.
There you have also a changelog documenting the differences. Steve Underwood
commit here the latest changes in spandsp source.
http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
Kristijan
2012/1/11 Olivier
Hello Paul,
this is a second backtrace taken from a core dump file later on. See
attachment.
Kind regards,
Jonas.
On 01/17/2012 03:59 PM, Paul Belanger wrote:
On 12-01-17 05:16 AM, Jonas Kellens wrote:
Hello list,
where can I post the output of the trace taken from a file :
Why not try include into startup/down init script of asterisk ?
Regards,
On Tue, Jan 17, 2012 at 9:41 AM, Bryant Zimmerman brya...@zktech.com wrote:
I have written a program that monitors asterisk to make sure my peers and
channels are all in good order. The program calls asterisk once a min
(Solved)
From: Luis Morales faston...@gmail.com
Sent: Tuesday, January 17, 2012 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] /etc/init.d script and calling asterisk
On Tue, 17 Jan 2012, Bryant Zimmerman wrote:
I have solved my issue. Danny and several others suggested that it might
have something to do with environment settings when running form the
init.d folder and that is what it turned out to be.
Just to be overly pedantic, your issue had nothing to
I have a Grandstream HT-502 device connected to my Asterisk PBX. It is
configured not to place anonymous calls, and from my mostly layman
reading of the invitation that the device sends, it should not be
anonymous. However, the Asterisk PBX sends an anonymous invitation to
our SIP trunk
No, in fact, they can't answer any of the phones in the ring-group.
On Tuesday 17 January 2012 11:04:35 am Eric Wieling wrote:
Can they answer the call by pressing the line key when simply picking up
the handset does not answer the call? If so, then the users are not
properly seating the
I did try asterisk 10.1-rc1 and spandsp-0.0.6pre18 and everything
seems to work OK, without teaking config.
Congrats to all !
I didn't know about this changelog.
Thanks for telling about it.
2012/1/17, Kristijan Vrban vrban.l...@googlemail.com:
I use the latest spandsp source from the
- Original Message -
I use the latest spandsp source from the freeswitch git.
There you have also a changelog documenting the differences. Steve
Underwood
commit here the latest changes in spandsp source.
http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp
Does
Hi,
I wasn't aware of the difference in quality between landline and
mobile phones, or that cellphones use a low bit rate.
I did a test again, landline voice quality is better.
From what you observed, how much drop in quality do I expect when
switching from ulaw to g729 for a normal
On 18-01-12 04:57, Roi Stork wrote:
Hi,
I wasn't aware of the difference in quality between landline and
mobile phones, or that cellphones use a low bit rate.
I did a test again, landline voice quality is better.
From what you observed, how much drop in quality do I expect when
switching from
Hi Zohair,
By using only asterisk it's not possible. So used progremming languages and
do realtime billing at your ends.
like 1st caller will take complete amount ($5) and if 2nd call will come
then deduct used amount and share remaining amount to others like that.
On Tue, Jan 17, 2012 at 9:54
Hi,
I understand this, but I think there isn't any option that helps us to
reduce cost while call is in progress.
One option that I was thinking is to check elapsed time by core show
channel channel-id and deduct the amount but we need to check it every
second or x seconds via AMI.
Regards,
Batter is used DB to store intime of call then when ever currect used time
is required then deduct from intime - current time.
On Wed, Jan 18, 2012 at 1:01 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Hi,
I understand this, but I think there isn't any option that helps us to
reduce
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