Hello,
Asterisk version 11.13.1
I'm trying use realtime meetme application with c and q option. If both options
are used together in meetme table under 'opts' field, c option (Announce
user(s) count on joining a conference.) doesn't work i.e. system doesn't play
user counting to caller. Is it
Date: Thu, 5 Sep 2013 12:11:36 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] high cpu average load
On Thu, 5 Sep 2013, Kamlesh Kumar wrote:
Running one asterisk server with below details.
Only SIP to SIP calls. No real time
Hello,
Running one asterisk server with below details.
Only SIP to SIP calls. No real time configuration, no recording, no voicemail,
no IVR, no codec translation. Average CPU load varies between 4 to 30 for 150
to 200 concurrent calls and we start getting problem in call quality like delay
Hi,
Asterisk 1.6.2.9
PHP 5.3
Mysql 5.0
Can anyone suggest hardware specification for 500 hundred concurrent SIP only
calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching.
There is only one requirement is to execute one php script on call hangup (h
extension) which
] limitation on number of contexts in
extensions.conf
On Friday 26 July 2013, Kamlesh Kumar wrote:
Thank you Carlos,
you were right, there was one empty file among all included files which
were causing this problem.
Couple of more queries:
Will system performance be affected
in
extensions.conf
On Wed, Jul 24, 2013 at 11:49 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello
Asterisk version 1.6.2.9.
I want to know is there any limitation on number of contexts or including
external file (#include filename) which can be defined in extensions.conf.
When I
Hello
Asterisk version 1.6.2.9.
I want to know is there any limitation on number of contexts or including
external file (#include filename) which can be defined in extensions.conf.
When I try to include around 40 external files, my dialplan doen't get reloaded.
Regards,
Kamlesh
/congested at this time (0:0/0/0)
Regards,
Kamlesh
Date: Tue, 4 Jun 2013 10:27:11 -0500
From: mr...@imminc.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar wrote:
SIP.conf
[100]
username=100
secret=password
type=friend
: 103 BYE
Content-Length: 0
Regards,
Kamlesh
Date: Fri, 31 May 2013 08:50:38 -0500
From: mr...@imminc.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar wrote:
Yes that's correct, when I use u-law call works fine
proceedings.
Regards,
Kamlesh
Date: Wed, 29 May 2013 08:42:39 -0500
From: mr...@imminc.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
Kamlesh Kumar wrote:
Call even doesn't go to the ITSP. I tried without AGI script and the results
were
Hello Matthew,
Call even doesn't go to the ITSP. I tried without AGI script and the results
were same.
Regards,
Kamlesh
Date: Tue, 28 May 2013 18:32:19 -0500
From: mr...@imminc.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
Of Kamlesh Kumar
Sent: Monday, May 27, 2013 2:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G.729 codec in pass-thru mode
Hello,
Trying to use g729 in pass-thru mode.
Call flow:
SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729) When
using G.729, call
Hello,
Trying to use g729 in pass-thru mode.
Call flow:
SIP IP Phone (G.729)--Asterisk(1.6.2.9)---SIP Trunk to ITSP(G.729)
When using G.729, call is not getting connected. Below is the extract from CLI.
== Using SIP RTP CoS mark 5
-- Executing [12127773456@default:1] AGI(SIP/100-,
Hello, We are using around 100 real time sip peers with phpagi. On asterisk
cli, getting frequent message 'failed to extend from 512 to 676'. Once we
execute 'sip reload', this message disappear for some time and then comes back.
Please let us know the solution for this. asterisk version
2013/3/7 Steve Edwards asterisk@sedwards.com
Please don't top-post.
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
How about a description of what you are
Hello, We need to setup asterisk server for 1000 extensions and in this setup
only extension to extension dialling is required (without call recording and
voicemail), like intercom calling. Please let us know what can be the best
economic solution/setup for this. Thanks,Kamlesh
On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
We need to setup asterisk server for 1000 extensions and in this
setup only extension to extension dialling is required (without call
recording and voicemail), like intercom calling. Please let us know what
can be the best economic solution/setup
extensions
On Thu, 7 Mar 2013, Kamlesh Kumar wrote:
Technology is SIP and asterisk is not handling the media, what is
cheapest solution to be used for SIP client.
Client? How about a free SIP softphone?
Server? How many calls per second? How many simultaneous calls? Any
half-way recent box
, Kamlesh Kumar wrote:
softphone is not going to be used in this setup. Hardphone is required.
Around 60-70 simultaneous calls would be required.
OK. So figure on about 6 UDP packets, about 3.5 KB per call. Not a big
deal.
I'd look for a reliable system and build 2 so you can swap between
Hello, Please suggest the way to change the time zone in below sip debug logs.
INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From:
xx sip:xxx...@xxx.xxx.xxx.xxx;tag=as23a29r59To:
header
you can use asterisk function SIP_HEADER(name). If you want to permanently
change date why not change system date/time?
Regards,
-Qasim
On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
Please suggest the way to change the time zone in below sip
Hello, Asterisk Version 1.6.2.9 on below hardware. We are using 100 Realtime
SIP extensions. CPU : 1 x Intel® Core-i5 3.3 GHz.
RAM : 4 GB DDR-3 SDRAM
Hard Disk : 500 GB Hard Disk For last few days, getting below messages on
asterisk cli. We googled to find the solution for this but could not
Hello, Trying to install libpri version 1.4.11.3 on Centos 5.5. but getting
below errors. Please suggest the resolution. gcc -Wall -Werror
-Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD -MT copy_string.o
-MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c
gcc -Wall
No, I'm trying first time. thanks,Kamlesh
From: asterisk_l...@earthshod.co.uk
To: asterisk-users@lists.digium.com
Date: Mon, 30 Jul 2012 11:22:54 +0100
Subject: Re: [asterisk-users] libpri error
On Monday 30 July 2012, Kamlesh Kumar wrote:
Hello, Trying to install libpri version
: Mon, 30 Jul 2012 11:49:30 +0100
Subject: Re: [asterisk-users] libpri error
(Do not write anything before the original message. The proper place for a
reply is *after* the thing you are replying to.)
On Monday 30 July 2012, Kamlesh Kumar wrote:
From: asterisk_l...@earthshod.co.uk
To: asterisk-users@lists.digium.com
Date: Mon, 30 Jul 2012 11:58:44 +0100
Subject: Re: [asterisk-users] libpri error
(Do not write anything before the original message! The proper place for a
reply is *after* the thing you are replying to.)
On Monday 30 July 2012, Kamlesh Kumar wrote:
when I issue
still waiting for valuable reply on this. Regards,Kamlesh
From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 15 Jun 2012 12:19:48 +
Subject: [asterisk-users] voicemail password with phone instrument
Hello,
voicemail password is not getting changed through
Hello, voicemail password is not getting changed through phone handset while
IVR indicates that password has been changed. During google I found that
uniqueid column must not be changed so it is not changed. Please guide on this.
During debug log I found below but in mysql db new password
...@setcolombia.com
Hi, try some like this:
[PERL snippet using get_data AGI command]
On Tue, 22 May 2012, Kamlesh Kumar wrote:
I tried it but it doesn't work.
beep file gets played, and when I enter any digit(s), it doesn't get
stored in $keys variable.
1) Does enabling AGI debugging
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension
not found. #!/usr/bin/php -q
?phpinclude_once (phpagi-2.14/phpagi.php);
include_once (/phpagi-2.14/phpagi-asmanager.php);
$agi = new AGI();
$as = new AGI_AsteriskManager();
$exten =
Hi, can we load the settings of /etc/dahdi/system.conf from database table in
real time. thanks,Kamlesh --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
at 04:56 +, Kamlesh Kumar wrote:
Hello,
For realtime configuration, in /etc/asterisk/extconfig.conf file, what
should be the family name to configure general sip.conf parameters.
family name = driver,database name,table name
thanks,
Kamlesh
From: i...@pack-net.co.uk
Hi,
I need to configure global parameters in sip.conf like rtptimeout,
rtpholdtimeout, rtpkeepalive, domain, session-timers etc... in real time
architecture. Please suggest the way to do it.
thanks,
Kamlesh --
Hello,
Using AGI script to accept the input from caller but input value is not getting
stored in variable.
Extract from AGI Script:
$agi = new AGI();
$agi- exec('Background','press_one0press_two0press_zero0');
$agi- exec('Read','NUMBER,,1,3');
$agi- verbose (You have entered.$NUMBER);
Hello,
# rpm -qa | grep kernel
kernel-headers-2.6.18-274.18.1.el5
kernel-PAE-2.6.18-128.el5
kernel-devel-2.6.18-274.18.1.el5
kernel-PAE-devel-2.6.18-274.18.1.el5
[root@localhost ~]# uname -i
i386
Trying to install dahdi-linux-complete-2.3.0.1+2.3.0 on CentOS but get below
error. Can you
,
Kamlesh
Date: Wed, 15 Feb 2012 14:39:00 +0200
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] error during dahdi installation on centos
On Wed, Feb 15, 2012 at 12:11:18PM +, Kamlesh Kumar wrote:
Hello,
# rpm -qa | grep kernel
To: asterisk-users@lists.digium.com
From: t...@softins.co.uk
Date: Fri, 2 Dec 2011 12:27:19 +
Subject: Re: [asterisk-users] DIALSTATUS Values
In article snt142-w54267269808afd17bccd5891...@phx.gbl,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
In addition to my reply:
I used
Hello,
I need to capture the DTMF digits dialled by user on current connected calls
and store them in variable.
scenario:
Manual Call Transfer:
User A dialed to B
B answered the call and want to transfer the call to user C manually. User B
dials *2 to get the ring tone again and then
which will let you continue in the DIal-plan after
the Dial command on hangup.
Regards,
Sammy.
On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
I'm using AGI scripting with asterisk and need to execute certain commands just
after Dial(). But once
,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
In addition to my reply:
I used to fetch the value using print_r function but that also tells that
there is no value
in data section.
$dialstatus=$agi-get_variable(DIALSTATUS);
print_r($dialstatus);
SIP/10036-00b8AGI Rx GET
Hello,
I'm using AGI scripting with asterisk and need to execute certain commands just
after Dial(). But once dial command is executed, further commands/instructions
are ignored.
$agi-exec(Dial,SIP/100);
$dialstatus = $agi - get_variable(DIALSTATUS);
if($dialstatus[data]==ANSWER)
Regards,
Sammy
On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
asterisk version 1.6.2.7
I want to get the start time of all active calls from console, could you please
let me know the best way to get it.
thanks,
Kamlesh
for an ongoing bridged call-initiator side only.
Regards,
Sammy
On Wed, Dec 14, 2011 at 1:16 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
'sip show channel' also does not give this info.
sip show channel f600ed29f561d57
localhost*CLI
* SIP CallI
Curr. trans. direction
Hello,
asterisk version 1.6.2.7
I want to get the start time of all active calls from console, could you please
let me know the best way to get it.
thanks,
Kamlesh --
_
--
Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts
using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI
script, I get empty value.
Extracts from AGI Script:
!!?
Regards,
Sammy.
On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts
using PHP. I'm unable to retrieve
Values
Can you also paste the Asterisk Console logs around the part where AGI is
dialing and after the dialing part ! make sure AGi debug is enabled as well.
On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hello,
in /etc/extension.conf
[privoip]
exten = _00X
-w45a64e4743de653da591...@phx.gbl,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
I tried to search the answer of my problem but unable to get resolution so
sending to you
guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm
unable to
retrieve the DIALSTATUS value
2011 11:44:34 +
Subject: Re: [asterisk-users] DIALSTATUS Values
In article snt142-w45a64e4743de653da591...@phx.gbl,
Kamlesh Kumar kamlesh_...@hotmail.com wrote:
I tried to search the answer of my problem but unable to get resolution so
sending to you
guys. I'm using asterisk 1.6.2.7
49 matches
Mail list logo
