Could it be distro-related?
I have various versions of asterisk (from 1.4 upto 11.3) running
paravirtualized or HW-virtualized with XEN.
Normally i use the pre-build packages from suse, only when i want to try
a release-candidates i need them myself.
hw
-Original Message-
From: Sandeep
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from two
side. but it is not good for me because 200 is the name of extension and
when i config asterisk
Hi Hans,
If we use the pre-built packages on say ubuntu (my server os), can i enable
options like when i do when i compile and do a menuselect? I mean can i
enable the cdr odbc, del odbc etc modules that I need?
On Tue, Apr 23, 2013 at 1:13 PM, Hans Witvliet aster...@a-domani.nl wrote:
Could
@Hans,
Now I feel its distro related as I am getting the same error when I try to
compile and run asterisk 1.8.. what distro are you using? I think I need to
change the distro I'm running on..
On Tue, Apr 23, 2013 at 1:44 PM, Sandeep Raju sandeepr...@practo.comwrote:
Hi Hans,
If we use the
@Hans, I just tried installing from pre-built packages (which has asterisk
1.8). Its working fine! :) only the compiled installed versions were
giving me the error!..
PS: sorry for spamming with multiple mails..
On Tue, Apr 23, 2013 at 2:10 PM, Sandeep Raju sandeepr...@practo.comwrote:
I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in
server. I am having some issue in audio quality. I want to enable jitter
buffer on asterisk but don't know, how to do. Any one can help me.
--
_
-- Bandwidth
Good morning,
We recently fell back to the most recent build of asterisk 1.8 down from 11.3
and I believe we've crossed some sort of limit for 1.8. Our dialplan is 515723
entries long with 6263 distinct contexts. Both are loaded realtime via odbc
(mysql). Previously at the end of a dialplan
Hi all,
I'm wondering what the recommendations are for using music on hold on
asterisk. As far as I understood from various pages on the web and a
response from the IRC channel, I am to avoid using mp3 files because of
licensing and transcoding issues. correct?
I am currently using asterisk
On Tue, Apr 23, 2013 at 02:17:47PM +0530, Sandeep Raju wrote:
@Hans, I just tried installing from pre-built packages (which has asterisk
1.8). Its working fine! :) only the compiled installed versions were
giving me the error!..
PS: sorry for spamming with multiple mails..
Distro packages
@Tzafrir,
I uninstalled the version 11.2 and compiled the version 1.8.12.2 as
mentioned in that page... its working fine now.. as my virtual machine was
running on KVM.. i think i faced the same issue mentioned in that issue
report..
I even went further and uninstalled 1.8.12.2 and install
my gcc version is as follows
gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3
On Tue, Apr 23, 2013 at 6:12 PM, Sandeep Raju sandeepr...@practo.comwrote:
@Tzafrir,
I uninstalled the version 11.2 and compiled the version 1.8.12.2 as
mentioned in that page... its working fine now.. as my virtual
On Tue, Apr 23, 2013 at 02:30:24PM +0200, Frederic Van Espen wrote:
Hi all,
I'm wondering what the recommendations are for using music on hold
on asterisk. As far as I understood from various pages on the web
and a response from the IRC channel, I am to avoid using mp3 files
because of
On 04/23/2013 03:12 PM, Tzafrir Cohen wrote:
If you use that mode, you're probably doing something wrong following an
ancient guide.
Well, these modes are the ones documented in the sample conf files that
came with asterisk 1.8.13.0:
snip
; valid mode options:
; files -- read files from a
On 2013-04-23 08:47, Sandeep Raju wrote:
my gcc version is as follows
gcc (Ubuntu/Linaro 4.6.3-1ubuntu5) 4.6.3
On Tue, Apr 23, 2013 at 6:12 PM, Sandeep Raju
sandeepr...@practo.com wrote:
@Tzafrir,
I uninstalled the version 11.2 and compiled the version 1.8.12.2 as
mentioned in that
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my calls
for the office. the problem is the management of the CDRs. i can see the
master.csv file, but it is not very friendly for the secretary of this
office to
Well, the question is, what your secretary wants to do. Only see the
CDRs or more? Realtime? One simple method would be to mail her the
CSV-File, so she can open it with Excel or Calc (Open Office).
Am 23.04.2013 16:35, schrieb aristidis tsitras:
Hi. i am running asterisk in a low powered
That would be nice.
is there a way to have it ready in xls?
if yes, then i could send it put a cron to send it every night/week/month.
On 04/23/2013 05:42 PM, Thorsten Göllner wrote:
Well, the question is, what your secretary wants to do. Only see the
CDRs or more? Realtime? One simple
On Tuesday 23 April 2013, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my calls
for the office. the problem is the management of the CDRs. i can see the
master.csv file, but it is not
On 23/04/2013 11:09 AM, A J Stiles wrote:
On Tuesday 23 April 2013, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my calls
for the office. the problem is the management of the CDRs. i
On 04/23/2013 06:23 PM, Ron Wheeler wrote:
On 23/04/2013 11:09 AM, A J Stiles wrote:
On Tuesday 23 April 2013, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my
calls
for the office.
On 23/04/2013 11:42 AM, aristidis tsitras wrote:
On 04/23/2013 06:23 PM, Ron Wheeler wrote:
On 23/04/2013 11:09 AM, A J Stiles wrote:
On Tuesday 23 April 2013, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
On Tue, 2013-04-23 at 17:35 +0300, aristidis tsitras wrote:
Hi. i am running asterisk in a low powered machine (alix2d13 from
pcengines) without any gui. the machine works fine to route all my calls
for the office. the problem is the management of the CDRs. i can see the
master.csv file,
- Original Message -
From: Brandon Mackie bmac...@awktane.com
We recently fell back to the most recent build of asterisk 1.8 down
from 11.3 and I believe we’ve crossed some sort of limit for 1.8.
Our dialplan is 515723 entries long with 6263 distinct contexts.
Both are loaded
After struggling with one way audio issues as a result of STUN binding
errors on both the Asterisk side and the Chrome side, we've decided to just
simply go with a TURN relay for RTP packets until the issues are resolved.
I configured rtp.conf so that all of the STUN related entries are commented
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