Hey Guys,
I am trying to set a time limit on my calls which are produced by sending
the following information to asterisk manager port.
Is there any solution to set time limit somewhere? for example 'Channel' =>
SIP/Provider/0|60|L(3)?
'Channel' => SIP/Provider/0
'Context' =>
Dears,
There is a PSTN line shared between 2 asterisk servers, (openvox 4FXO
lines)
The outgoing call of the one server may be conflict with the established
call of the other one,
is any way to force the Asterisk or Dahdi to dial after hearing the Dial
tone ?
--
Pezhman Lali
Dear,
would you please let me know how to recognize DTMF during recording? for
example escape digits ? in asterisk 1.6.2.X?
..
$agi-record_file(a,gsm,1234567890);
..
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Dear
is any problem if the max of callgroup 63, in source code ?
best
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some PRI providers block the callerid, and you can use only your head number
as cid.
best
Pezhman Lali
http://blog.lopl.net
On Sun, Aug 14, 2011 at 4:40 AM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 13 Aug 2011, bilal ghayyad wrote:
I need that if five IP Phones make outside
-users mailing list
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Dear all
may be it isn't related .
but I want to shared my VOIP experiences in my new weblog.
http://blog.lopl.net
Help me to improve it by your comments and ideas.
Best
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Dear
for having an stable system which limit option is good for ulimit comand ?
2-is any option for making asterisk crash-free?
Best
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New
is it possible to prevent 100% cpu usage by asterisk, with ulimit?
On Wed, Aug 10, 2011 at 11:53 AM, Pezhman Lali l...@lopl.net wrote:
Dear
for having an stable system which limit option is good for ulimit comand ?
2-is any option for making asterisk crash-free?
Best
--
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to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Dear,
which one is more powerful and more stable(openh323 and ooh323) for
h323-sip proxy?
Best
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Dear
is it possible to send ring(call) to all devices with same (sip_username) in
all servers ?
in this schematics, some bodies have shared lines. so all lines must be in
service .
Best
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hours, with only 10 calls, the cpu went more than 100% , and
crashed.
the bt full result of gdb was attached
I have some questions now,
1-is any problem in the attached report.
2-does asterisk 1.4 more stable than 1.6 in this case?
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mailing list
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New to Asterisk
for disabling, like IAX and SIP in realtime
mode?I had the same experience with IAX, when our online users grew up,
asterisk was crashed. but by disabling rtcache, we had better condition
best
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:
http://www.asterisk.org/hello
asterisk-users mailing list
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Dear,
do you have any successful experience for installing SHT-8C/PCI/FAX (synway)
with asterisk ?
is it compatibe with asterisk (dahdi/zaptel)?
best
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thanks,
this delay is occurred on asterisk server, between dial execution and
CALLED .
On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse...@selbytech.com wrote:
On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net wrote:
Dear
I have a small pbx with asterisk 1.6.2.16.
I have
Dears
thanks for your all helps.
with assigning ip to the sip.conf and disabling srv_lookup the delay was
removed.
thanks again
best
On Tue, May 10, 2011 at 12:31 PM, mahesh katta maheshka...@flexydial.comwrote:
Dear Pezhman Lali,
Just below lines add in you sip.conf, after this if you get
Dear
I have a small pbx with asterisk 1.6.2.16.
I have a funny problem, there is exactly 40sec between dial execution and
sending first invite packet on sip.
do you have any idea where the problem is ?
Best regards
--
Pezhman Lali
check your odbc connection with isql
best
On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.com wrote:
You're using 1.4.2. Why not try upgrading to a more recent release of 1.4
(I believe 1.4.41 is current) and see if your issue has been resolved.
Thanks,
--Warren Selby,
may be the ip phone has the problem, try reset as factory
On Fri, Apr 29, 2011 at 8:03 PM, Mike l...@net-wall.com wrote:
What I am looking for? Here is a snippet, with some info obfuscated. I can
see the bad request, but why there is such a message isn’t obvious.
--- SIP read from
Dear
try phpagi. it has a lot of useful functions.
in this scenario you will lose your digit, set a check point between each
digit gathering
best
On Wed, Apr 27, 2011 at 6:17 PM, David asterisk@spam.lublink.netwrote:
Hi,
Consider the following situation :
SIP/asterisk-001dAGI Rx
check this
http://www.voip-info.org/wiki/view/Asterisk+sip+nat
On Thu, Apr 21, 2011 at 2:12 PM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear * users,
in your opinion, when using a * as a public server, is good practice
enabling nat=yes in sip.conf for all the peers?
Can
check this url, let me know if any problem
http://www.voip-info.org/wiki/view/Asterisk+video
http://www.voip-info.org/wiki/view/Asterisk+video
http://www.voip-info.org/wiki/view/Asterisk+videobest
On Thu, Apr 21, 2011 at 9:00 PM, Steve Davies davies...@gmail.com wrote:
Hi,
Can anyone let
yes, ami is your unique answer.
what is msisdns ?
On Wed, Apr 13, 2011 at 3:18 PM, Albert alber...@wp.pl wrote:
Hi,
I am working on integration of 2 systems: asterisk and messaging platform.
What I need is to access somehow information about current calls. Should I
do it over AMI ?
I
Dear
there is some problem.
the true way for running php script, is using agi not system.
second after 5 sec, a lot of channel variables were removed, it makes your
program wrong.
with some little experience you can add your script to a2billing, try it.
best
On Sat, Apr 9, 2011 at 7:22 PM, Bruce
extension even if it was only run in x extension.
Regards,
On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali l...@lopl.net wrote:
Dear
there is some problem.
the true way for running php script, is using agi not system.
second after 5 sec, a lot of channel variables were removed, it makes your
for your network it's optional to receive the fax on your server, you can
pass the received fax to the destination, like a voice call with g711 and no
VAD.
ask if you need more info.
best
On Wed, Apr 6, 2011 at 4:55 PM, Bert Van Kets mail...@vankets.com wrote:
On 1/04/2011 13:04, Khaled W.
fail2ban(opensource) is a good choice for you
best
On Wed, Apr 6, 2011 at 1:16 PM, Gordon Henderson gordon+aster...@drogon.net
wrote:
On Tue, 5 Apr 2011, Steve Edwards wrote:
On Tue, 5 Apr 2011, Gilles wrote:
I'm no expert of iptables, and it seems like it can handle banning
IP's
using the realtime functions for voicemail solve this problem.
you can insert a query from your agi to add new voicemail box.
is it what you need ?
On Tue, Apr 5, 2011 at 10:17 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 5 Apr 2011, vip killa wrote:
Is it possible to create a
you can not see what you send, change the config in the mailing list options
On Sun, Mar 6, 2011 at 6:36 AM, sean darcy seandar...@gmail.com wrote:
I can't seem to send anything. Let's see if this shows up.
--
_
--
I think a2billing is the best billing opensource system, but try astbill,
new url http://astbss.org/
http://astbss.org/but if you want to setup a large system select
enterprise system, these systems are useful for small and med networks.
best
On Sat, Mar 5, 2011 at 8:56 PM, bilal ghayyad
Dear
this note is only for fresh administrators don't think about asterisk
security.
I found fail2ban very useful for anti asterisk hacking, so I want to share
it with fresh admins.
some hackers try your sip or iax2 ip with a lot of username/password, may be
after 1 million try, one
hi
using database as realtime functions solves your first problem,
for second try by using dns
best
On Mon, Feb 28, 2011 at 1:54 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
I would like to have two Asterisk machines to have redundancy between them,
so if first machine failed then we
help?
Regards
Bilal
--- On *Mon, 2/28/11, Pezhman Lali l...@lopl.net* wrote:
From: Pezhman Lali l...@lopl.net
Subject: Re: [asterisk-users] Two Asterisk machines for redundancy
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: bilal ghayyad
dear
I have a good exp in setting up 79xx on sccp, with sccp-b library, and tftp
server, which part is the main problem for you?
best
On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
Hi,
Anyone who
some outside sip provider does not accept dtmf,
if you have not this problem in your local, ask your outside carrier
best
On Wed, Feb 16, 2011 at 7:27 AM, asterisk asterisk aster...@ck-lee.comwrote:
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I
really it's too difficult to understand, please explain more clear
On Tue, Feb 15, 2011 at 5:17 AM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Hi,
How can I configure my asterisk server so that I can receive incomming
calls comming from the same IP from where my server also receives
you can run any function in your hangup extension,
exten = h,1,...
best
On Tue, Feb 15, 2011 at 12:21 PM, Richard Zheng rzh...@gmail.com wrote:
Hi,
In ACD queue, is it possible for the agent to take some actions when the
caller hangs up? For example, to let the agent to enter some
please send your sip.conf, is any NAT procedure implemented in your network?
On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Hi,
I manage an SBC which stands between my company server farm and some SIP
telco trunks. The system works fine, for inbound
this command will not work.
what is your main purpose?
do u need to have a conference with a group of sip phones?
best
On Tue, Feb 15, 2011 at 3:13 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
I am wondering if its possible to have sometime like this:
exten 100 = Dial
I know there is not a good place for ask this question. but I can not find
in other ways.
Dear,
Do you have any experience for changing the logo of cisco 7905 on sccp
firmware?
best
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Dear
I had good experience with asterisk + spandsp for sending and receiving
fax, if your ip phone supports fax, you need asterisk only as g711(no vad)
gateway.
best
On Sun, Feb 13, 2011 at 7:00 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/12/2011 10:53 PM, Mark Willis wrote:
Is it
as you know you have 2 ways. using ami or .call files. if you
have experience, the AMI is more powerful.
you must have a context in your extensions.conf to manage agent procedures,
it looks like a simple context, that you must have, for managing queues.
with .call file or ami dial your customers,
Dear
is any way to have a secure (encrypted) rtp line between cisco 79XX and
asterisk with SCCP?
best
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Dear
a2billing also provided call_back daemon, try it
best
On Sun, Feb 6, 2011 at 12:57 AM, Paul Belanger pabelan...@digium.comwrote:
On 11-02-05 06:07 AM, Gilles wrote:
I'd like to configure Asterisk so that...
1. I ring it from my cellphone with CID number displayed, just to
notify
Dear,
Meetme is a default conference application, but you can try conference or
konference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference
http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference
Dear,
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
best
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sorry for no url
https://code.google.com/p/faxter/
https://code.google.com/p/faxter/best
On Sun, Jan 30, 2011 at 12:51 PM, Pezhman Lali l...@lopl.net wrote:
Dear,
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
best
check your /etc/asterisk/asterisk.conf and post it here
best
On Sat, Jan 29, 2011 at 2:22 PM, Gilles codecompl...@free.fr wrote:
Hello
On a uClinux-based appliance, ps aux shows multiple Asterisk
processes:
380 root 11990 S asterisk -f
381 root 11990 S asterisk -f
383
= apache
;astctl = asterisk.ctl
[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6
On Sat, Jan 29, 2011 at 4:32 PM, Gilles codecompl...@free.fr wrote:
On Sat, 29 Jan 2011 15:47:53 +0330, Pezhman Lali l...@lopl.net
wrote:
check your /etc/asterisk/asterisk.conf and post it here
Here goes:
root
don't forget to install spandsp, and replace the value of Channel with true
value.
best
On Fri, Jan 28, 2011 at 4:26 PM, bakko asannu...@gmail.com wrote:
Hello,
you have to use a callfile
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Create a callfile, for example test.txt,
did u compile lib_pri ?
On Thu, Jan 27, 2011 at 7:30 PM, William Stillwell
will...@stillwellsoft.com wrote:
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error:
Should have only transmitted 0 frames!
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405
Dear Please send us, your iax configurations.
best
On Mon, Jul 5, 2010 at 7:10 AM, bruce bruce bruceb...@gmail.com wrote:
Hi guys,
I have two Asterisk servers (with FreePBX) connected together with IAX2
trunking. When I call from server A-B call connects but hangs up after 30
seconds. What
add the a2billing configurations to the sip.conf
best
On Thu, Jul 1, 2010 at 7:34 PM, bruce bruce bruceb...@gmail.com wrote:
Yes, you are missing a whole bunch of configurations from creating SIP
users to making sure they show as peers on Asterisk to making sure you use
dnid, etc.You
please send your extension.conf
2010/6/30 Anahi Ludueña a_ludu...@hotmail.com
Hi people,
we have some extensions which are included in the IVRs and/or queues.
Everything works fine, but the calls done from these extensions are hang up
after 30 o 35 seconds. If they are not included in the
Dear,
some iax phones,(with built in router) have problem, with our asterisk
server, there is no way sound if they call out, but it's ok if somebody
calls them.
the normal iax phones without router have'nt ny problem.
can u help me?
the version of kernel is 2.6.18 and asterisk is 1.4.26.2
Best
Dear,
some iax phones,(with built in router) have problem, with our asterisk
server, there is no way sound if they call out, but it's ok if somebody
calls them.
the normal iax phones without router have'nt ny problem.
can u help me?
the version of kernel is 2.6.18 and asterisk is 1.4.26.2
Best
Dear,is any way to find silent channels , and disconnect them after 30 secs?
best
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is any program , to manage freemin on sim cards ,for gsm gateways that
connected to the asterisk, for termination?
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asterisk-users mailing list
To UNSUBSCRIBE or
I have problem with packet size of voip packets, in a big network.
what is the best monitoring tools and analyzer for this purpose?
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asterisk-users mailing list
To
by using rtcachefriends=yes it was done.
--- On Sat, 1/31/09, Pezhman Lali pezhman_l...@yahoo.com wrote:
From: Pezhman Lali pezhman_l...@yahoo.com
Subject: [asterisk-users] iax clients were unregistered after 30sec
To: asterisk-users@lists.digium.com
Date: Saturday, January 31, 2009, 7:34
Dear,
Our iax clients's ip and port in the database were removed automatically, after
30 secs.
the iax info is saved in odbc and postgresql .
asterisk=# select * from iax_buddies where username='9706015';
name | username | type | secret | md5secret | dbsecret | transfer |
inkeys |
Dear,
I added new field to cdr table , named service and type varchar(20),
but in extensions.conf with the following command, nothing to be saved.
exten = _X.,1,Set(CDR(service)=OUT)
does asterisk support this ability ?
is any setting must be changed, before that ?
best
Mani
To: Asterisk Users asterisk-users@lists.digium.com
Date: Monday, January 26, 2009, 1:18 PM
Pezhman Lali schrieb:
I added new field to cdr table , named
service and type varchar(20),
but in extensions.conf with the following command,
nothing to be saved.
exten = _X.,1,Set(CDR(service)=OUT
Dear,
the goto function to the iax dialing, makes bill duration and call duration
wrong, in cdr.they are equal to ringing time.
the cdr will be produced and saved into the dbase, when the callee picks up the
phone.
is any way to have real duration time ?
[main]
exten =
Dear,
because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on
asterisk.
I can not use goto , because of some limitations.
is any way to decrease it?
Best,
[MAIN]
exten = _12X.,Dial(LOCAL/${ext...@test/n,60)
[TEST]
exten _X.,1,Dial(${ext...@next_gateway,60)
Dear,
I have combined asterisk 1.4 with cisco 2600 connected to PRI,
the biggest probelm is that, the cisco does not send busy her sip_486 to
asterisk, for busy callee .
can u help me to find the solution?
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asterisk 1.2 , is enough old to make a lot problems,
upgrade to 1.4 or 1.6 and enjoy it.
integration opensips( ser) and astersik, is the best solution for the big voip
systems.
--- On Sat, 12/27/08, Mike Trest m...@trest.com wrote:
From: Mike Trest m...@trest.com
Subject: Re:
Dear,
is any way to change the iax packets?
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Dear,
the sip phones that registered, in to the asterisk 1.4.x have the echo in their
callings to pstn.
how this echo can be canceled?
Best
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@lists.digium.com
Date: Thursday, November 20, 2008, 12:01 PM
Pezhman Lali wrote:
Dear,
the sip phones that registered, in to the asterisk 1.4.x have the echo
in their callings to pstn.
how this echo can be canceled?
H - you don't give much to go on...
What is the connection to the PSTN (i.e
is any command , shows the current rate of each channel?
--- On Mon, 11/10/08, Kristian Kielhofner [EMAIL PROTECTED] wrote:
From: Kristian Kielhofner [EMAIL PROTECTED]
Subject: Re: [asterisk-users] changing the size of voice packets
To: Asterisk Users Mailing List - Non-Commercial Discussion
Dear Fateme
two good refrences:
http://articles.techrepublic.com.com/2415-1035_11-94140.html
and
http://www.trixbox.org/forums/vendor-forums-certified/sangoma/solved-sangoma-101d-card-trixbox-asterisk-1-4-19-1
hope to help u
best
Pezhman
--- On Tue, 11/11/08, fateme fatah [EMAIL PROTECTED]
mp3player, is just for your need,
use it this like
exten = _X.,1,mp3player(http://www.test.com/test.mp3;)
try this page
http://www.voip-info.org/wiki-Asterisk+cmd+MP3Player
best
--- On Wed, 11/12/08, Singer X.J. Wang [EMAIL PROTECTED] wrote:
From: Singer X.J. Wang [EMAIL PROTECTED]
Subject:
, November 10, 2008, 3:00 PM
Hi!
On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED] wrote:
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because
of bandwidth failure.
You can specify size of voice
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because
of bandwidth failure.
thanks in advance
Mani
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Dear,
compiling appconference 2.0. with g729 enabled, makes the quality of voices too
low,
for low voices , there is'nt any problem, but normal voices have alot of noises.
best
Mani
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Dear,
is any command to show the codecs of channels , in asterisk 1.4?
Best
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solved
with
sip show channels
best
--- On Wed, 9/17/08, Pezhman Lali [EMAIL PROTECTED] wrote:
From: Pezhman Lali [EMAIL PROTECTED]
Subject: [asterisk-users] codec of channels
To: asterisk-users@lists.digium.com
Date: Wednesday, September 17, 2008, 5:42 PM
Dear,
is any command to show
Dear,
I have a little problem with app_conference,
the very low power voices, were amplified, too much,
and normal voices were destroyed.
codec=g729
asterisk=1.4.19
app_conference =last released
best
Mani
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Dear,
I have some limitations to install zaptel because of kernel reinstalling.
also there is'nt any zaptel device installed in the server.
but I need to install meetme, for conferencing .
can u help me ?
Best
Mani
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Of Pezhman
Lali
Sent: September-11-08 5:59 AM
To: asterisk
Subject: [asterisk-users] meetme without zaptel
Dear,
I have some limitations to install zaptel because of kernel reinstalling.
also there is'nt any zaptel device installed in the server.
but I need to install meetme
Dear,
do u have any idea to playback a remote file (with url address) ?
for example :
exten = _X.,1,playback(http://www.test.com/test.gsm;);
best
Mani
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AstriCon 2008 -
Dear,
is any solution for replacing .call files into the database?
best
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
Dear,
Is any configuration for using outgoing via database(realtime)?
Best
Mani
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
your mail is not clear at all.
if you want to change the path of sendmail ,do this with mailcmd, in the
voicemail.conf,
if you want to send a voicemail to a class of emails, using dbase is more
easier.
let me to know more, about your problem.
--- On Sun, 6/29/08, fateme fatah [EMAIL
Dear,
I am using ser + asterisk, for outgoing calls,
my problem is that the session was not closed if the caller says bye.
can u help me ?
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AstriCon 2008 - September 22
Dear,
your hardware is good for more than 200-300 calls,
configure asterisk for more details in debug,
the output in console is more useful.
also plz attach your main configurations for conference,
viewing consumed ram and cpu during conference, can help
--- On Mon, 6/16/08, fateme fatah [EMAIL
using odbc+( postgres or mysql) is more stable,
but at all odbc + postgres is recommended
--- Sherwood McGowan [EMAIL PROTECTED]
wrote:
Steve Prior wrote:
Tilghman Lesher wrote:
Correct; it's actually a workaround for a bug in
the MySQL drivers. It was
discovered long after 1.2
Dear,
is any test for using iax-phone with asterisk in larg
system?
for example cpu-users, ram-users, cpu-call,
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Dear,
after a lot of searching and testing I can not find a
total solution for nat, with ser -- asterisk.
now I have 3 selections:
1)using iax-phones instead of sip phones with asterisk
2)using sip phones registered in asterisk,
3)using sip phones with ser/openser and, searching for
new ways,
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