Hi All,
I want to make a new file of CLI log everyday. So I just make a shell
script in asterisk log directory. My file is working fine and making new
file with the name of *full_2012-01-27*. But strange I noticed that
asterisk is updating my newly crested files even i don't reload asterisk.
So h
asterisk jobs писал 27.01.2012 06:49:
> Hello everyone, I
have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
>
> Chain fail2ban-ASTERISK (1 references)
> num target prot opt source destination
> 1 DROP all -- 0.23.20.189
0.0.0.0/0 [1]
> I al
This is how I use a wav to mp3 script on Mixmonitor in my dialplan
(Asterisk 1.8.7.0).
...
same => n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
^{FILENAME})
...
and my script is...
#!/bin/bash
WAV="/var/spool/asterisk/monitor/$1"
MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
MP3DEST="/var/sp
On 1/25/2012 10:29 AM, Faraj Khasib wrote:
I am trying to convert files that are .wac to mp3 after mixmonitor
> command is called but it doesnt execute the command, I tried the
command in terminal it worked, any help please ... below is my dial
> plan
what version of asterisk are you using ?
Hello everyone,
I have noticed getting wired IPs blocked by Fail2ban. Has anyone else seen
this or can explain this?
Chain fail2ban-ASTERISK (1 references)
num target prot opt source destination
1DROP all -- 0.23.20.189 0.0.0.0/0
I also get things like, 0.
Are you by chance using templates (!) In your sip.con? Ive had access denied
errors befor when running as non root.
- Original message -
> I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood
> of:
>
> WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL l
On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards
wrote:
> The OP was using MIXMONITOR_EXEC (although I wonder about the '&&' syntax)
> so he doesn't need to explicitly execute (via system()) his commands.
Wow. Never knew that was possible. I still don't like the syntax, but
good to know.
For optim
I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood of:
WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic
error or missing database
AFAIK, I'm not doing any database puts (or gets). There were no such
warnings in 1.8.8.0.
What do I need to do to sil
On Thu, 26 Jan 2012, David Backeberg wrote:
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg wrote:
shebang /path/to/bash
PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg wrote:
> shebang /path/to/bash
>
> PATH=$1
> lame --arguments $1.wav $1.mp3
> if [ -f {$1}.mp3 ] ; then
> rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug. (gotta close the i
On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib wrote:
> Hello Guys,
> I am trying to convert files that are .wac to mp3 after mixmonitor command is
> called but it doesnt execute the command, I tried the command in terminal it
> worked, any help please ... below is my dial plan
> exten=6500,n,Se
On Thu, 26 Jan 2012 10:35:14 -0700
Mike Diehl wrote:
> While trying to track down a T.38 issue, I came across a series of log
> entries like this:
>
> [Jan 26 10:23:31] WARNING[32508]: udptl.c:948
> ast_udptl_new_with_bi
Dietmar Zlabinger wrote:
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Am 26.01.2012 18:43 schrieb "Steve Edwards" :
> On Thu, 26 Jan 2012, eherr wrote:
>
> It is accessible from HTTP.
>>
>> However, the access list only allows access from my home and the password
>> is strong.
>>
>
> Can you configure it to 'syslog' accesses where you can monitor it.
>
Hi
For this scenario you can use the group and group_count functions and create
a hint dialplan like this:
exten => trunkname,hint,custom:trunkname
When you reach the maximum number of available channels set the hint in use:
Set(DEVICE_STATE(Custom:confcorso)=INUSE)
To remove:
Set(DEVICE_STA
On Thu, 2012-01-26 at 18:21 +0100, Patrick Lists wrote:
> On 26-01-12 18:08, Jeff LaCoursiere wrote:
> [snip]
> >
> > I'm also very interested in working examples, especially if someone has
> > set it up for SIP termination "trunks" rather than Dahdi.
>
> Maybe I am missing something here but why
On Thu, 26 Jan 2012, eherr wrote:
It is accessible from HTTP.
However, the access list only allows access from my home and the
password is strong.
Can you configure it to 'syslog' accesses where you can monitor it.
Maybe your access lists are invalid, misunderstood or not being honored.
--
Hi all,
While trying to track down a T.38 issue, I came across a series of log
entries like this:
[Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr:
Unable to allocate socket: Too many open files
[
Just a WAG, but I'm guessing they may have a limited number of lines and
don't want one phone hogging 2-3 at a time.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Thursday, January 26, 2012
On 26-01-12 18:08, Jeff LaCoursiere wrote:
[snip]
I'm also very interested in working examples, especially if someone has
set it up for SIP termination "trunks" rather than Dahdi.
Maybe I am missing something here but why would you want to emulate a
keysystem with analog (thus single call) li
It is accessible from HTTP.
However, the access list only allows access from my home and the password is
strong.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hayes
Sent: Thursday, January 26, 2012 10:
On Thu, 2012-01-26 at 10:48 -0600, Tim Nelson wrote:
> - Original Message -
> > On 01/26/2012 09:46 AM, Tim Nelson wrote:
> > > Greetings-
> > >
> > > I currently have a customer that *requires* key-system functionality
> > > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
- Original Message -
> On 01/26/2012 09:46 AM, Tim Nelson wrote:
> > Greetings-
> >
> > I currently have a customer that *requires* key-system functionality
> > in an Asterisk PBX. On a SIP phone, the BLF keys need to show the
> > current state of the analog lines attached to the system (DA
On 01/26/2012 09:23 AM, Russell Brown wrote:
I'm using Manager API Originate to initiate calls from SIP channels (via
phpagi FWIW) and it all works well except
...the CallerID for the SIP channel specified in users.conf isn't set for
the call :-(
If I explicitly set the Callerid in the Man
On 01/26/2012 07:22 AM, Vieri wrote:
Hi,
I was receiving fax calls just fine until recently. I'm now having random
disconnections.
Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends
it to a iaxmodem (exten 10025 below). All's apparently as expected except for
the
On 01/26/2012 09:46 AM, Tim Nelson wrote:
Greetings-
I currently have a customer that *requires* key-system functionality in an
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of
the analog lines attached to the system (DAHDI FXO). By pressing one of these
keys (for
Greetings-
I currently have a customer that *requires* key-system functionality in an
Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of
the analog lines attached to the system (DAHDI FXO). By pressing one of these
keys (for line 1 for example), the dialed number needs
On 20/01/12 01:36, eherr wrote:
It is also register on an AudioCodes MP-118.
Thanks,
-E
Is the Audiocodes gateway accessible online? Have you set a strong
password for it's web interface (and cli if it has one)? It is possible
someone is breaking into that and getting the SIP password o
I'm using Manager API Originate to initiate calls from SIP channels (via
phpagi FWIW) and it all works well except
...the CallerID for the SIP channel specified in users.conf isn't set for
the call :-(
If I explicitly set the Callerid in the Manager Originate API call then
it works but the A
Hello
how are you?
Can you give me advice on which are the best free or not (free prefered)
that use SIP Transfer.
Thanks a lot!!!
--
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New to Asterisk? Join u
Hi,
I was receiving fax calls just fine until recently. I'm now having random
disconnections.
Faxes are received over an ISDN BRI line and Asterisk 1.4 detects it and sends
it to a iaxmodem (exten 10025 below). All's apparently as expected except for
the fact that the following message comes u
Il 25/01/2012 22:52, Michael Keuter ha scritto:
Outcry! :-)
I'm outcrying too :)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 01/25/2012 11:10 AM, Ishfaq Malik wrote:
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
I confirm that Chan
On 01/13/2012 06:58 PM, Administrator TOOTAI wrote:
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting
2012-01-26 10:11, Eyal skrev:
>
> Thanks
>
>
>
> But this is not what I am looking for, in this way I can start the
> sound file from some point in the file but the callers must hear the
> file until the end.
>
> I need something that allows me to start from some place in the file
> and end it in
You can use a combination of ChanSpy() and a local extension playing the
required file to caller/callee.
On Thu, Jan 26, 2012 at 2:11 PM, Eyal wrote:
> Thanks
>
> ** **
>
> But this is not what I am looking for, in this way I can start the sound
> file from some point in the file but the cal
Thanks
But this is not what I am looking for, in this way I can start the sound
file from some point in the file but the callers must hear the file
until the end.
I need something that allows me to start from some place in the file and
end it in some other place in the file (say song from time
check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback
Nasir Iqbal
ICTBroadcast
SMS, Fax and Voice broadcasting solution
http://www.ictbroadcast.com/
On Wed, Jan 25, 2012 at 8:29 PM, Eyal wrote:
> Hi,
>
> How can I play a sound file from the middle and end it after a
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