On 14/05/2010, Motiejus Jakštys desired@gmail.com wrote:
Talking about file permissions, on Linux everything is possible using
POSIX ACLs. You can set specific rights to files/directories for
certain users.
Note 1: if setting group permissions is enough, use that.
Note 2: Asterisk and
The voip-info.org tells it differently :
*MYSQL(Clear* ${resultid}*)*
Frees memory and data structures associated with result set.
*MYSQL(Disconnect* ${connid}*)*
Disconnects from named connection to MySQL.
But it does not make a difference...
It's strange that every mysql-query is the same
The only thing that makes it ring in step 3 (so after the queue) is
calling the Queue-command with the r-option.
So there is no music on hold but a ringtone, when the caller sits in the
queue.
Now the question is: when I want to use music on hold while inside the
queue, how can I get the
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong ? becasue i
can't excute it successfully.
please open the following url to see my code:
Jonas Kellens wrote:
But it does not make a difference...
I'm running Asterisk 1.4.x, what version are you running?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
Hi!
Issue solved.
Looks like all I was missing was one parameter:
fromuser=
That's interesting - could be related to this:
http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html
You were probably caught be the fact that you are using extension numbers
also as SIP user
It does not make a difference because it is the same result : All the
other queries go well, just the last one gives this 'WARNING'.
Using 1.4.25.1
Jonas.
On 05/14/2010 11:46 AM, Doug Lytle wrote:
Jonas Kellens wrote:
But it does not make a difference...
I'm running Asterisk
- Original Message -
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong ? becasue i
can't excute it successfully.
please open the following url to
Jonas Kellens wrote:
It does not make a difference because it is the same result : All the
other queries go well, just the last one gives this 'WARNING'.
You may want to give func_odbc a try, several say it's a better way:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc
I've been trying to get the hang of Agents and Queues and I must say
its a little unclear as to how things work.
So maybe someone has some better idea
From what I can work out an Agent is meant to be nothing more than a
virtual device that can be moved around physical devices... by login
--- On Thu, 5/13/10, Zoa zoach...@securax.org wrote:
Can you try trunk = no ?
Lifesaver...
trunk=no made the interference go away.
I have clean audio now.
Quote: IAX Trunking needs support of a hardware timer.
I'm supposing my system is using the DAHDI-driven Digium cards on my
i've also tied this tests:
- changed hardware
- upgrade to 1.4.31
- kernel recompiled with 1000 Hz option
- changed SO (Slackware 13)
- run the system on hardware (no ESXi)
But i've not resolved the problem.
Do you have any idea?
On Thu, May 6, 2010 at 11:54 AM, nik600 nik...@gmail.com wrote:
--- On Fri, 5/14/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
You were probably caught be the fact that you are using
extension numbers
also as SIP user names for your phones (here: 3666). This
is not a good
thing to do, better use an alphanumeric username or
The way I reproduce it is not simple. I am receiving calls on 1.6.0.27
and doing FastAGI scripting. Then it happens after a few hours of steady
calls. I'm looking for a simpler way to reproduce it.
Looks like issue 0017321 describes the bug in one way. This is a serious
bug, I am checking to
I think that the clock resets would cause no audio or garbled audio
every 20 minutes, not constant interference.
Could you tell us how many simultaneous calls were in the trunk and what
the size is of 1 voice packet ?
Can you try putting maximum 30 calls per trunk (use multiple trunks if
Hi,
If I am expecting too much here, please just tell me so, but I was
under the impression that this was put into 1.6.x
I have 2 types of SIP devices. For argument's sake, let us say that
one type of device can talk G722 and ALAW, and the other only talks
ALAW. I have directmedia=yes.
Calls
On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote:
Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp
0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax.
Hint: you need to install spandsp then run ./configure then make menuselect
:)
I was able to
Hi,
I'm using dynamic realtime with asterisk 1.6.0.24, I'm having a strange
problem with queue_members...
If I update only 'membername' field on queue_members table asterisk won't
refresh the change, but if I update another field like interface everything
works as expected, I've found this
On Fri, 14 May 2010, Vieri wrote:
I'm supposing my system is using the DAHDI-driven Digium cards on my
motherboard. I don't know how hardware timers work and if Digium
hardware rely on the motherboard (my system clock is going too fast and
my ntpd is constantly adjusting the clock by -2.6
I'm getting the following message in my full log at startup and my
realtime sip peers aren't being found. My realtime extensions have no
errors. The table sippeers exists in the database. Is this a known
problem?
res_config_mysql.c: Table sippeers not found in database. This table
should
Steve, Thanks for the heads up, after extensive testing today, I have
decided to edit t30.c to mark as debug, as I recorded a call between two
analog fax machines using mixmonitor, and noticed the same waveform
patterns, as a call into spandsp/receivefax.
I must admit, I am way happier with
On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote:
I'm getting the following message in my full log at startup and my
realtime sip peers aren't being found. My realtime extensions have no
errors. The table sippeers exists in the database. Is this a known
problem?
res_config_mysql.c:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
During tests with a Grandstream GXP280 phone, I found that pressing
'hold' button, the other extension (Qutecom softphone) is put on hold
but without music. Then, when resuming the conversation, I listen the
other user again but he/her no
On 05/14/2010 04:17 PM, Tilghman Lesher wrote:
On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote:
I'm getting the following message in my full log at startup and my
realtime sip peers aren't being found. My realtime extensions have no
errors. The table sippeers exists in the database.
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