running with root user.
On Tue, Nov 29, 2011 at 1:52 AM, Edwin Lam edwin@officegeneral.comwrote:
On 11/24/11 2:13 AM, virendra bhati wrote:
I did the same as you mention like that
*echo -n 1234 | md5sum
another things to check are:
- the permission of the file /tmp/pass.txt
Hi Edwin,
I did the same as you mention like that
*echo -n 1234 | md5sum
*
On Thu, Nov 24, 2011 at 3:13 AM, Edwin Lam edwin@officegeneral.comwrote:
On 11/22/11 9:02 PM, virendra bhati wrote:
On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati virbh...@gmail.com
mailto:virbh
asterisk-users mailing list
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Hi List,
I want to change the asterisk flow. right now call startd from
extensions.conf. Is there any way by which we can changed it to
extensions.ael or extensions.lua ?
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Virendra Bhati
+91-9172341457
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in sip.conf [default] section or for each sip user
decalred who needs to start call in context defined in AEL/LUA?
** **
Regards,
Gohar
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
*Sent
extension (bhati-test, 111, 2) exited non-zero on
'SIP/2218-0664'
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Hi List,
I want to use text file to get password information with Authenticate
Application. I am using asterisk 1.6.2.11. I made text file at
/tmp/pass.txt with below information.
*pass.txt*
Virendra: 81dc9bdb52d04dc20036dbd8313ed055
Vijay : 9996535e07258a7bbfd8b132435c5962
Virendra Bhati
://www.asterisk.org/hello
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(SPYGROUP=spy)
exten = 43681156,n,NoOp(***${SPYGROUP})
exten = 43681156,n,ChanSpy(DAHDI,g(spy))
exten = 43681156,n,Hangup()
when I used chanspy without option then It works
like Chanspy(DAHDI)
Any help will be appreciated
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Virendra Bhati
+91-9172341457
Software Engineer
only used these channels not find all
channels.
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Virendra Bhati
+91-9172341457
Software Engineer
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Hi List,
Is there any way by whcih I can make group of user as per my requiremt and
start spy on these channels whic Chanspy ?
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+91-9172341457
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hi
as you know meetme default recording file format is wav file. you may change
is too gsm for reduce file size.
or if you want then you may use monitor or mixmontor for gsm recording too.
On 6 Oct 2011 12:09, mahesh katta maheshka...@flexydial.com wrote:
Thanks for reply,
This recording is
hi,
you are using pattern matching and not using the right syntax
like that.
exten = _520,1,answer
like that.
On 5 Oct 2011 21:47, salaheddine elharit salah.elharit...@gmail.com
wrote:
Hello list
i have one question related to meetme,i have to providers with the first
one
i put the
Virendra Bhati
+91-9172341457
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overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:
Hi Sam,
You are right. I am looking for the same
listen to the playbacks.
Regards,
Sammy.
On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.comwrote:
Hi List,
I make a script for .call file and then I started playback on local
channel but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n
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rfc2833
and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.
Kristijan
2011/9/13 virendra bhati virbh...@gmail.com:
Hi
1st check that how many manager is connected into the server. 1 or more
then
you can say that 2 DTMF is capture by asterisk for same events.
manager
Hi List,
Is there any way by which I can broadcast any audio file to all members into
the conference ?
I don't want to play file individual channels.
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+91-9172341457
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Konference. So I required more then 1 AMI connection might be 1 connection
for 1 konference. Because I will play some IVR files to get DTMF and on this
DTMF i will check the correct DTMF. So that I will get the right user with
correct input.
So please guide me.
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Virendra
to each of them
On Wed, Sep 7, 2011 at 15:59, virendra bhati virbh...@gmail.com wrote:
Hi list,
I want to know that will it be possible that more then 1 AMI is connected
from single Linux machine with different name ?
As we know that default 1st AMI connection will come with 127.0.0.1
:
Though this might have been resolved/accomplished already but I've couple
of questions for Virendra Bhati.
1- If you are doing this to make new accounts for new users, why couldn't
you use Asterisk realtime(DB) based configurations of
Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than
]
and over-write that part only. If a new user then just append. This way file
data loss will be minimized(may even avoided totally).
Those were all my suggestions, if anyone else can add valuable comments to
this.
-
sammy
On Mon, Sep 5, 2011 at 11:45 AM, virendra bhati virbh...@gmail.comwrote
Hi Sammy,
Yes I am asking about AstDB only.
On Mon, Sep 5, 2011 at 2:00 PM, Sam Govind govoi...@gmail.com wrote:
Are you talking about AstDB or MySQL as DB backend for asterisk?
On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati virbh...@gmail.com wrote:
Hi Sammy,
Thanks for share your
Hi Raza,
Thanks , but there is no ned of Sip.conf and extensions.conf files.
As Daniel refered the web page which is enough for all the tasks
On Sat, Sep 3, 2011 at 5:18 PM, Daniel Tryba dan...@tryba.nl wrote:
On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
Please guide me
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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: bhavik\r\n);
fputs($socket, mode: files\r\n);
fputs($socket, directory: /var/lib/asterisk/moh\r\n);
fputs($socket, Reload: yes\r\n);
fputs($socket, ActionID: 9873497149817\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
?
After doing all no success :((
-
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Virendra
...@gmail.com wrote:
Why php? Isn't vi the only way?
On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati virbh...@gmail.com wrote:
Hi list,
I want ot do basic work (add-edit-delete) into asterisk configuration
files,
like sip.conf, manager.conf,musiconhold.conf etc.
Please guide me how
Hi List,
How to play wav files to all konference members at a time. I want to play
with the help of AMI connection.
I have tested that we can play channel base file playing. But it will take
too much time if users are more then 20
-
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Virendra Bhati
+91-9172341457
receiving anything(fgets) :
See this page
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP will
help you.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24
: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
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Virendra Bhati
+91-9172341457
Software Engineer
.
http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
*Sent:* Thursday, August 25, 2011 4:02 PM
*To:* Asterisk Users Mailing List
Virendra Bhati
+91-9172341457
Software Engineer
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: warning: format ‘%ld’ expects type ‘long int’, but
argument 6 has type ‘sf_count_t’
app_espeak.c:458: warning: format ‘%ld’ expects type ‘long int’, but
argument 7 has type ‘sf_count_t’
make: *** [app_espeak.o] Error 1
On Mon, Aug 22, 2011 at 6:05 PM, virendra bhati virbh...@gmail.com wrote:
Hi
options visit:
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Hi,
In CLI please press Konference then Tab from keyboard then you will see all
the command of Konference.
You may use AMI connection for batter usw.
On 3 Aug 2011 19:46, Danny Nicholas da...@debsinc.com wrote:
You need to provide more information - is line in SIP or DAHDI, what
release
of
past 15 minutes
%l4 Process fraction (processes running / total processes)
%l5 The most recently allocated pid
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right ?
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Virendra Bhati
+91-9172341457
Software Engineer
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, Jul 4, 2011 at 7:13 PM, Earl e...@micpc.com wrote:
On Monday, July 04, 2011 05:10:43 AM virendra bhati wrote:
[RecordPrompts]
exten = ,1,Answer()
exten = ,n,NoOp(WelCome to conference section)
exten = ,n,Playback(ConfDemoWC)
exten =
,n,MixMonitor(tmp/00Record/
On Thu, Apr 21, 2011 at 11:31 AM, virendra bhati virbh...@gmail.com wrote:
hi,
Hint will work all VoIP hardware or specific hardware device ?
I am planing to using CISCO 79XX series so please suggest me..
And What about softphone ?
On Wed, Apr 20, 2011 at 8:57 PM, Danny Nicholas da
Hi List,
Do you have any suggestion in DTMF case ?? I have change my sangoma card to
digiumbut still same..
On Mon, May 23, 2011 at 4:08 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
After changes relaxdtmf=no in chan_dadhi.conf. problem is not resolve
On Mon, May 23, 2011
help
me so thatI will make asterisk as per my need.
-
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Virendra Bhati
+91-9172341457
virbh...@gmail.com
Software Engineer
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at 5:13 AM, virendra bhati virbh...@gmail.com
wrote:
Hi List,
I have installed Kannel server into my Linux server. I have asterisk
installed into the same server. Now I want to connect both opensource
project. As per the VoIP-info website I read that in asterisk there is
an
option
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?
On 06/16/2011 01:52 AM, virendra bhati wrote:
Hi List,
I want to secure my server from the hacker's. What is the case by
which I can protest it.
I have done security of Dialplan, Sip,IAX base security. For linux we
are working on Iptables. What else is left so that I will do it too
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options visit:
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+91-9172341457
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and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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http
Hi List,
I want to secure my server from the hacker's. What is the case by which I
can protest it.
I have done security of Dialplan, Sip,IAX base security. For linux we are
working on Iptables. What else is left so that I will do it too...
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+91
Hi List,
Is there any way by which we can remove asterisk from machine without
deleting folder manually? I did google and gets various solution by no
success. even after deleted asterisk will be there .
-
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Virendra Bhati
+91-9172341457
Asterisk Engineer
(CALLERID(name)=Virendra Bhati)
But when call reach to destination number then only number is display, name
was display as *unknown *
Is this issue of voip provider or Asterisk 1.6.2.18 ?
I contact them they replay me that it's your end issue not my end.
-
Thanks and regards
Virendra
has its own commands (yum, apt-get ecc)
Alex
*Da:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Per conto di *virendra bhati
*Inviato:* venerdì 10 giugno 2011 11:26
*A:* Asterisk Users Mailing List - Non-Commercial Discussion
*Oggetto
.
--
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Rajnikant Vanza
Call : +91-9737456583
Software Engineer
---
Working On Linux,C/C++,Asterisk Technology
Gandhinagar - Gujarat
On Thu, Jun 9, 2011 at 12:13 AM, virendra bhati virbh...@gmail.comwrote:
Hi List,
When we make
-users@lists.digium.com
On 2011-06-05 19:54, virendra bhati wrote:
Hi John Wilfer,
Thanks for replay. Now all things is working on asterisk 1.6.2.18 version.
But When I try the same thing on Asterisk 1.4.X then facing problem.
Is this the problem of ControlPlayback 's option fields
Thanks Paul,
Link was too awesome. I read and check all related command too.
Thank you for your help.
On Wed, Jun 8, 2011 at 2:37 AM, Paul Belanger pabelan...@digium.com wrote:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any
, Krishna Sumanth Chava ksch...@gmail.comwrote:
Hi Virendra,
Set DTMF option in the Makefile to 1 and then recompile/install the
app_konference module.
Thanks
Krishna
On Tue, Jun 7, 2011 at 1:31 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I am trying to get DTMF into conference
Hi,
I am using CentOS 5.6 and I am getting error message
In my case old command is find.
On Wed, Jun 8, 2011 at 5:25 PM, Karsten Wemheuer k...@gmx.de wrote:
Hi,
Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger:
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List
will be the VoIP calling call flow in Incoming and outgoing calls?
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)=XXX) in dialplan.
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is the problem in this case please help me..
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registered into
asterisk server.
But thanks you clear my concept into Voip Call routing too.
On Thu, Jun 9, 2011 at 12:15 AM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 8 Jun 2011, virendra bhati wrote:
I have working experience of asterisk with PRI lines. Recently I have took
VoIP
Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
-
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+91-9172341457
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Hi List,
I am trying to get DTMF into conference room. for conference I am using
Konference module. Konference don't have an option of DTMF gets. Is there
any way by which I can get DTMF within conference room?
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
creating more issue into conference when you are working on DTMF base
services.
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Virendra Bhati
+91-9172341457
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) ?
Is there any way by which we will implement like by upload ControlPlayback
from asterisk 1.6 to 1.4 or else ?
ControlPlayback(filename[,skipms[,ff[,rew[,stop[,pause[,restart[,options]]])
On Sun, Jun 5, 2011 at 2:16 PM, Johan Wilfer li...@jttech.se wrote:
On 2011-06-04 13:38, virendra bhati wrote
sleep(2) function..
and I can't do any work in between this sleep time into server.
On Thu, May 26, 2011 at 7:00 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Thursday 26 May 2011, virendra bhati wrote:
Hi ,
Thanks for reply ..
What is the meaning of that line which you have
.
ControlPlayback(${filename},6,3,1,*#2456790,,,o(${position}))
Please put some light on these too.
On Wed, Jun 1, 2011 at 1:50 AM, Johan Wilfer li...@jttech.se wrote:
On 2011-05-30 14:32, virendra bhati wrote:
Hi List,
Asterisk 's *ControlPlayback* will used for play any recorded file
or update options visit:
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Thanks a lot all,
Now my view is clear ...
On Sun, May 29, 2011 at 3:15 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sun, 29 May 2011, virendra bhati wrote:
Hi List,
I have stupid question but I want to know it. Why we use the PRI insted of
BRI ? Just for the sake of number
,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
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+91-9172341457
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Hi List,
I have stupid question but I want to know it. Why we use the PRI insted of
BRI ? Just for the sake of number of lines or any thing else ?
And why SIP is used for making calls rather then IAX? Even we know IAX takes
1 channel for making calls?
-
Thanks and regards
Virendra Bhati
it's not work.*
exten = h,n,NoOp(***never come into execution
**)
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+91-9172341457
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not work.*
exten = h,n,NoOp(***never come into execution
**)
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+91-9172341457
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, the author was illustrating that you must
include the full path to the binary
Sent from my iPhone
On May 26, 2011, at 8:07 AM, virendra bhati virbh...@gmail.com wrote:
Hi ,
Thanks for reply ..
What is the meaning of that line which you have mention on the recent
conversation
System(path
How to make outgoing calls from DID and what is theway to get incoming calls
from DID.
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don't
allow you to dial out. So it all just depends on what is allowed.
On May 26, 2011 1:56 PM, virendra bhati virbh...@gmail.com wrote:
How to make outgoing calls from DID and what is theway to get incoming
calls
from DID.
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+91
Hi List,
Is it possible to put your call on hold after bridge with another call ? I
want to use asterisk dialplan not softphone button or IP-phone device.
If yes then how to get back on bridge mode of on-hold call again ?
-
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+91-9172341457
Asterisk
the field
rtchachefriends in table sip_buddies. And exclude the field qualify from
sip_buddies. Set YES in field rtcachefriends.
Att,
Rafael Saraiva
2011/5/21 virendra bhati virbh...@gmail.com
Hi List,
After read the link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . I changes
]: rtp.c:1809 ast_rtp_read: Unknown RTP codec
126 received from '192.168.193.134'
-- Stopped music on hold on SIP/100-
*
On Sat, May 21, 2011 at 7:38 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Tue, May 17, 2011 at 10:16 AM, virendra bhati virbh...@gmail.com
wrote:
hi list
, virendra bhati virbh...@gmail.com wrote:
Hi List,
As per you suggestion I had made changes but still no improvements. Some
thing was missing so I am listing aging the configuration files details
here.
*res_mysql.conf*
[mpathsala]
dbhost = localhost
dbname = mpathsala
dbuser = root
dbpass
,
I am confuse about these CLI commands
*sip show users
sip show peers*
Can someone clear my doubt . what are the difference between them?
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+91-9172341457
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Hi James,
My question was related to CLI commands not sip.conf type
value(friends,peer,user)
main question was difference below commands
*sip show peers* Vs *sip show user*
On Mon, May 23, 2011 at 1:28 PM, virendra bhati virbh...@gmail.com wrote:
Hi James,
Thanks I give me the clear view
more because Events are getting with the help of Asterisk AMI not Konference
module..
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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Hi List,
After changes relaxdtmf=no in chan_dadhi.conf. problem is not resolve
On Mon, May 23, 2011 at 3:01 PM, Tony Mountifield t...@mountifield.orgwrote:
In article BANLkTin3dYqFxEfjUVkH+7HuXo_TLe_Z=w...@mail.gmail.com,
virendra bhati virbh...@gmail.com wrote:
I am using Asterisk 1.6.2.18
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
Hi list,
I am confuse about these CLI commands
*sip show users
sip show peers*
Can someone clear my doubt . what are the difference between them?
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Thanks and regards
Virendra Bhati
+91-9172341457
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Secret Accountcode
Def.Context ACL NAT
Why SIP/300 is not display here ??
Please help me I want to learn asterisk real-time concept to make my server
real-time.
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
:16, virendra bhati virbh...@gmail.com wrote:
hi list,
please help me how to know how many calls are on hold.
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Virendra Bhati
+91-9172341457
Asterisk Engineer
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Hi List,
How to put multiple call on hold by dialplan in asterisk?
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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New
hi list,
please help me how to know how many calls are on hold.
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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New
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
Hi list,
Is there any way by which we can put multiple calls into hold with asterisk.
like A to B.
then C to B and A on hold.
then D to B now C ,A on hold like wise..
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
Hi ,
I am using Cisco 7940/60 phone.
Is this okay or we need another phone for that. plz suggest me '
On Thu, May 12, 2011 at 3:52 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 12 May 2011, virendra bhati wrote:
Hi list,
Is there any way by which we can put multiple
webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Thanks and regards
Virendra Bhati
+91-9172341457
/mailman/listinfo/asterisk-users
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Thanks and regards
Virendra Bhati
+91-9172341457
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New to Asterisk? Join us for a live introductory webinar every
run:
/usr/sbin/asterisk -rnxpri show spans
/etc/init.d/asterisk status
/etc/init.d/mysql status
.
.
.
.
and so on!!
good luck!
Regards.
Juan.
Linux User #441131
On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati virbh...@gmail.comwrote:
Hi
How to know status of Asterisk,Mysql. PRI
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