Oh yes that will be more suitable but will still need to do it via AMI
Regards,
Zohair Raza
On Wed, Jan 18, 2012 at 11:35 AM, virendra bhati virbh...@gmail.com wrote:
Batter is used DB to store intime of call then when ever currect used time
is required then deduct from intime - current
Hi,
While compiling 1.8.8.1, I met the following error:
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o
I'm in the process of replacing an old server with a new one and are
making som changes in the infrastructure, the biggest change in my eyes
is moving from i386 to AMD64 arch. Yesterday I began migrating some
users from the old to the new server.
After only 57 concurrent calls in abount 13
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable the
loading of res_timing_timerfd.so and loaded res_timing_dahdi.so
instead:
noload = res_timing_timerfd.so
Can someone confirm that the nesting of macro's or the continuous and
simultaneous use of different macro's, can lead to stack-problems and
cause an Asterisk spontaneous reboot/restart ?
Kind regards,
Jonas.
On 01/17/2012 03:02 PM, Bryant Zimmerman wrote:
Jonas
From what I understand
Hi
have you open the port in rtp.conf ?
rtpstart=1
rtpend=2
On Wed, Jan 18, 2012 at 1:14 PM, shalu dhamija
shalu.dham...@rancoretech.com wrote:
Hello,
I am trying to deposit a voicemail message(using voicemail() application)
for a subscriber using asterisk-1.8.7.1. But i am
On Tue, Jan 17, 2012 at 08:12:04PM +0100, Kristijan Vrban wrote:
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.
Yes I've personally experienced issue with nested macros and eventually
asterisk failing to process call any further. So I moved onto using GoSUBs
and everything worked perfectly. Since then I'm using GoSUBs happily.
On Wed, Jan 18, 2012 at 4:54 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hi guys,
the problem was too many NATs on the way.
Although the server had a valid ip, it was behind a nat, as soon as I set
ip directly on the server, things worked fine.
Also, despite the huge delay, if the link has qos, the quality is very good.
On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind
Thank you for sharing your experience.
Anyone else ? Maybe one of the developers can confirm this risk of
working with macros ?
I'm now also moving towards GoSub where possible. I was experiencing
asterisk restarts every 40minutes. Exactly 40 minutes ! I've created a
backtrace of the core
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 18, 2012 6:36:35 AM
Subject: Re: [asterisk-users] Macro vs sub
Thank you for sharing your
Hi Arlen,
I'm interested in seeing what setup you settled on to get decent voice quality
over the Sat link? Which codec are you using, and what is the bandwidth usage?.
Are you doing just one concurrent call, Or multiple?.
-
Regards,
AJ Stanfield
- Original Message -
From: Arlen
On Wed, Jan 18, 2012 at 02:10:48PM +0200, Tzafrir Cohen wrote:
Nice to see he finally uses Git. Though it would be even nicer if it
were a stand-alone repo.
Almost there: https://gitorious.org/spandsp/spandsp-fs
But my script can, at this point, only fully create it from scratch and
not
on server side no special configuration is needed.
To have qos on the sat link, we contact sat link operator, and I think this
is the only way to do it.
The codec is g729. I´m not sure about the bandwidth, I think we have about
64Kbps allocated, because we almost don´t have concurrent calls.
The
I guess that's why people call it *disasterisk*
On Mon, Jan 16, 2012 at 8:10 AM, Vik Killa vipki...@gmail.com wrote:
Anybody? I've read this might be a deadlock
On Thu, Jan 12, 2012 at 8:09 AM, Vik Killa vipki...@gmail.com wrote:
Asterisk 1.6.1.22
On Thu, Jan 12, 2012 at 2:08 AM, Sammy
On 12-01-18 04:19 AM, Nyamul Hassan wrote:
Hi,
While compiling 1.8.8.1, I met the following error:
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o
On 01/18/2012 01:51 PM, Matthew Jordan wrote:
Anyone else ? Maybe one of the developers can confirm this risk of
working with macros ?
I don't think you need an Asterisk developer to tell you the risks of using
macros in deeply nested situations. Quoting the documentation of Macro:
Because
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable the
loading of res_timing_timerfd.so and loaded res_timing_dahdi.so instead:
noload = res_timing_timerfd.so
On 01/11/2012 02:39 PM, Olivier wrote:
Hi,
Maybe I missed it while checking it, but which spandsp version is
recommended to play with Asterisk 10 and T.38/T.30 gatewaying ?
I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here
(http://www.soft-switch.org/downloads/spandsp/)
Ah, apologies, I just re-read your given Asterisk version. Indeed,
I was using 1.8.5.0 at the time, not any 1.4.x release.
Any digium timing card will work as an OpenVZ compatible dahdi
timing device, I've seen this work on both Virtuozzo and OpenVZ.
Setting it up,
On Wed, Jan 18, 2012 at 02:52:47PM +0100, Johan Wilfer wrote:
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable the
loading of res_timing_timerfd.so and
Hi all,
I am installing Asterisk and Dahdi on my system and when I am installing dahdi
it tels me to install the sources of the 3.1 kernel.
How to do this on Arch?
Many thanks,
Christian
--
_
-- Bandwidth and Colocation
2012-01-18 16:45, John Knight skrev:
Ah, apologies, I just re-read your given Asterisk version. Indeed, I
was using 1.8.5.0 at the time, not any 1.4.x release.
Any digium timing card will work as an OpenVZ compatible dahdi timing
device, I've seen this work on both Virtuozzo and OpenVZ.
2012-01-18 17:50, Shaun Ruffell skrev:
On Wed, Jan 18, 2012 at 02:52:47PM +0100, Johan Wilfer wrote:
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by modifying modules.conf to disable the
On Wed, Jan 18, 2012 at 06:50:19PM +0100, Christian wrote:
Hi all,
I am installing Asterisk and Dahdi on my system and when I am installing
dahdi it tels me to install the sources of the 3.1 kernel.
How to do this on Arch?
On Debian: apt-get install linux-headers-`uname -r`
--
"Have you used 64 bit kernels (amd64) in your setup? Distribution?"
Aye, I use the current stable 64-bit rhel6 branch openvz kernel
with centos 6 on the node and scientific linux 6 in the template
without issue other than what I described before with
On 01/18/2012 12:15 PM, Johan Wilfer wrote:
2012-01-18 17:50, Shaun Ruffell skrev:
On Wed, Jan 18, 2012 at 02:52:47PM +0100, Johan Wilfer wrote:
2012-01-18 11:31, John Knight skrev:
Hi Johan,
I've run into a similar issue before. I didn't resolve the problem
per se, but I got around it by
On Wed, Jan 18, 2012 at 12:58:31PM -0600, Kevin P. Fleming wrote:
On 01/18/2012 12:15 PM, Johan Wilfer wrote:
2012-01-18 17:50, Shaun Ruffell skrev:
One question first though, is your new server able to keep accurate
time with nt, or is the clock drifting or experiencing heavy jitter?
Hi Paul,
The OS is CentOS release 6.1 (Final)
And, it is running kernel:
2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 19:48:09 GMT 2011 x86_64
x86_64 x86_64 GNU/Linux
Regards
HASSAN
On Wed, Jan 18, 2012 at 19:33, Paul Belanger pabelan...@digium.com wrote:
On 12-01-18 04:19 AM, Nyamul Hassan
On 01/18/2012 01:06 PM, Shaun Ruffell wrote:
On Wed, Jan 18, 2012 at 12:58:31PM -0600, Kevin P. Fleming wrote:
On 01/18/2012 12:15 PM, Johan Wilfer wrote:
2012-01-18 17:50, Shaun Ruffell skrev:
One question first though, is your new server able to keep accurate
time with nt, or is the clock
On 01/18/2012 01:44 AM, shalu dhamija wrote:
Hello,
I am trying to deposit a voicemail message(using voicemail()
application) for a subscriber using asterisk-1.8.7.1. But i am facing
aproblem in the rtp port allocation for a session due to which '488 Not
Acceptable' response is sent towards the
2012-01-18 20:06, Shaun Ruffell skrev:
On Wed, Jan 18, 2012 at 12:58:31PM -0600, Kevin P. Fleming wrote:
On 01/18/2012 12:15 PM, Johan Wilfer wrote:
2012-01-18 17:50, Shaun Ruffell skrev:
One question first though, is your new server able to keep accurate
time with nt, or is the clock
Hi, I'm seeing an odd issue at a recent installation and have been unable
to resolve it. Caller A calls Caller B and Caller B transfers to Caller
C. Using a blind transfer, if Caller C doesn't answer then Caller A gets
Caller C's voicemail. (as expected) However if doing an attended transfer
On 01/15/2012 08:14 PM, Louis Carreiro wrote:
Hey all!
I've been banging my head against the wall for a while (almost 18 hours
today alone) with this one... I migrated our incomming T1's from the
Option 11 to our Asterisk box this morning. We have 1 local T1 and 2
long distance T1's. The local
Hi Team,
Is there any way that asterisk can support Dialogic card, i have done lot
of search but could find any useful information.
Thanks
Vinod Dharashive
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com
Yes you may used Dialogic card with asterisk. but it's depends on the
requirements too.
On Thu, Jan 19, 2012 at 9:05 AM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi Team,
Is there any way that asterisk can support Dialogic card, i have done lot
of search but could find any useful
Hi,
I have not changed res_rtp_asterisk.c Its just that I have put the debug prints
in that file.
In asterisk 1.8.7.1 the allocation of rtp session is done in check_user_full()
function called from handle_request_invite. Since we are not handling the
authentication of the user I have
Hi, why don't you try write two macros only and recursively call both of
them incrementing a counter each time you call the inner macro. Also
print(NOOP) system stats along with the counter. You'll soon see what
happens.
The para Matthew quoted is cent percent true. But if you don't need to
call
Does Asterisk permit multiple registrations to the same host?
Each registration has a different username and password
The purpose is for billing, handling incoming calls is not important,
although it will be a bonus.
I guess I should also ask the converse, whether the receiving host can
accept
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