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?
-
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Virendra Bhati
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 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
ward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause)
:
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d php 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
wrote:
> On Thursday 26 May 2011, virendra bhati wrote:
> > Hi ,
> >
> > Thanks for reply ..
> > What is the meaning of that line
ch
lines in this application.
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 wrote:
> On 2011-05-30 14:32, virendra bhati wrote:
>
> Hi List,
>
> Asterisk 's *ControlPlay
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 wrote:
> On 2011-06-04 13:38, virendra bhati wrote:
>
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 Eng
x27;s creating more issue into conference when you are working on DTMF base
services.
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Virendra Bhati
+91-9172341457
Asterisk Engineer
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Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
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Virendra Bhati
+91-9172341457
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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 wrote:
> On 11-06-07 02:31 AM, virendra bhati wrote:
>
>> Hi List,
>>
>> Is there any way by which we can get the leng
g to get solution
On Tue, Jun 7, 2011 at 9:29 PM, Krishna Sumanth Chava wrote:
> 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, viren
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 wrote:
> Hi,
>
> Am Dienstag, den 07.06.2011, 17:07 -0400 schrieb Paul Belanger:
> > On 11-06-07 02:31 AM, virendra bhati wrote
What will be the VoIP calling call flow in Incoming and outgoing calls?
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)=XXX) in dialplan.
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at is the problem in this case please help me..
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ce already 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 wrote:
> On Wed, 8 Jun 2011, virendra bhati wrote:
>
> I have working experience of asterisk with PRI lines. Recently I have took
>
to 3.000
-- Executing [s@ivr-16:10]
Set("Local/13343757789@from-trunk-sip-wwisp1-7731;2",
"TIMEOUT(response)=10") in new stack
-- Response timeout set to 10.000
-- Executing [s@ivr-16:11]
Set("Local/13343757789@from-trunk-sip-wwisp1-7731;2",
"__IVR_RETVM=")
in..
> /Johan
>
> Original Message Subject: Re: [asterisk-users]
> ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan
> WilferTo: Asterisk Users Mailing
> List - Non-Commercial Discussion
>
>
>
> On 2011-06-05 19:54, vire
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
,n,Set(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.
---
urces: make uninstall
>
> distro: Every distro 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:* venerd
ctory 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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.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
&g
lman/listinfo/asterisk-users
>
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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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Virendra Bhat
gt;
> 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
>> a
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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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Hi Steve,
Thanks for share your knowledge. I will revert back to you after testing
with asterisk.
On Sun, Jun 19, 2011 at 6:46 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:
> On Sun, Jun 19, 2011 at 8:49 AM, Steve Totaro
> wrote:
> > On Sun, Jun 19, 2011 at 5:13
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 wrote:
> Hi List,
>
> After changes relaxdtmf=no in chan_dadhi.conf. problem is not resolve
>
> On Mon, May 23, 2011 a
On Thu, Apr 21, 2011 at 11:31 AM, virendra bhati 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 P
ve at disk.
Am I right ?
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Virendra Bhati
+91-9172341457
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, Jul 4, 2011 at 7:13 PM, Earl 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)
> &g
inutes
%l3 Load average over past 15 minutes
%l4 Process fraction (processes running / total processes)
%l5 The most recently allocated pid
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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" wrote:
> You need to provide more information - is line in SIP or DAHDI, what
release
> of Asterisk, etc.
>
>
erisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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+91-9172341457
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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 wrote:
> Hi List,
>
>
Hi List,
I want to know how many manager is connected into asterisk server. I have
made simple file but I don't have any idea how to get information back from
Asterisk CLI
Now how to get information into this PHP file
-
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Virendra Bhati
+91-9172341457
Sof
ny user is connected
> >
> > I’m not a php expert, but seems your php script is incomplete/ you are
> sending to socket (fputs) but note receiving anything(fgets) :
> >
> > See this page <
> http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP> will
>
cript is incomplete/ you are
>> sending to socket (fputs) but note receiving anything(fgets) :
>> >
>> > See this page <
>> http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP> will
>> help you.
>>
>> >
>> >
>> >
>>
intain a while loop and break loop on a condition toggled
> by web-page)
>
> See php section for other examples.
>
> http://www.voip-info.org/wiki/view/Asterisk+manager+Examples
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com
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
Hi List,
Any hint ?
Is there any posibility to stream played file os any chanels to all members
?
On Sat, Aug 27, 2011 at 6:03 PM, virendra bhati wrote:
> Hi List,
>
> How to play wav files to all konference members at a time. I want to play
> with the help of AMI connection
___
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>
.
*AMI login:- *
*login.php*
*AMI command:-*
Below commands are for musiconhold.conf. I want to add new MOH context into
it.
After doing all no success :((
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
wrote:
> Why php? Isn't vi the only way?
>
> On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati wrote:
> > Hi list,
> >
> > I want ot do basic work (add-edit-delete) into asterisk configuration
> files,
> > like sip.conf, manager.conf,musiconhold.conf etc.
>
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 wrote:
> On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
> > Please guide
is 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&#x
ile corresponding to one user i.e [user-1-area]
> 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.
>
> -
>
Hi Sammy,
Yes I am asking about AstDB only.
On Mon, Sep 5, 2011 at 2:00 PM, Sam Govind wrote:
> Are you talking about AstDB or MySQL as DB backend for asterisk?
>
>
> On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati wrote:
>
>> Hi Sammy,
>>
>> Thanks for share
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
er to each of them
>
> On Wed, Sep 7, 2011 at 15:59, virendra bhati 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
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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local extension will be like a conference
> member.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
> *Sent:* Monday, September 12, 2011 11:44 AM
> *To:* Asterisk Users Mailing Li
digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
> *Sent:* Monday, September 12, 2011 1:28 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] broadcast
>
> ** **
>
> Hi Ahmed,
>
main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173
>
>
> On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:
>
>> Hi Ahmed,
>>
>> Konference is
t; listener script should be able to tell you at the end of poll what user
> inserted with DTMF.
>
> So 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.
>
>
Biology/Que4)
> exten => s,n,playback(${p}/LQA/12/Biology/Que5)
> exten => s,n,playback(${p}/LQA/12/Biology/Que6)
> exten => s,n,playback(${p}/LQA/12/Biology/Que7)
> exten => s,n,Wait(10)
> exten => s,n,Hangup()
>
> This should work and konference should listen to t
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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gt; and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed.
>
> Kristijan
>
> 2011/9/13 virendra bhati :
> > Hi
> > 1st check that how many manager is connected into the server. 1 or more
> then
> > you can say that 2 DTMF is capture by asterisk
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idation:2] GotoIf("SIP/100-0034",
"0?notValidUser:ValidUser") in new stack
-- Goto (macro-Student_Validation,s,6)
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Virendra Bhati
+91-9172341457
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hi,
you are using pattern matching and not using the right syntax
like that.
exten => _520,1,answer
like that.
On 5 Oct 2011 21:47, "salaheddine elharit"
wrote:
> Hello list
>
>
>
> i have one question related to meetme,i have to providers with the first
one
> i put the number with 9 digit 520
hi
as you know meetme default recording file format is wav file. you may change
is too gsm for reduce file size.
or if you want then you may use monitor or mixmontor for gsm recording too.
On 6 Oct 2011 12:09, "mahesh katta" wrote:
> Thanks for reply,
>
> This recording is meetme conference record
nd chanspy only used these channels not find all
channels.
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Hi List,
Is there any way by whcih I can make group of user as per my requiremt and
start spy on these channels whic Chanspy ?
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Virendra Bhati
+91-9172341457
Software Engineer
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***)
exten => 43681156,n,Set(SPYGROUP=spy)
exten => 43681156,n,NoOp(***${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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ar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
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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
15:58:31] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.40'
[Nov 21 15:58:35] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec
126 received from '10.10.11.191'
[Nov 21 15:58:41] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP co
visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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at 9:47 PM, Warren Selby wrote:
> On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati wrote:
>
>> Hi,
>>
>> After deleting all space no improvements.
>>
>
> Try reversing the account code and password hash, like this:
>
> 81dc9bdb52d04dc20036dbd8313ed0
gards
Virendra Bhati
+91-9172341457
Software Engineer
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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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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.digi
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 wrote:
> On 11/22/11 9:02 PM, virendra bhati wrote:
>
>>On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati ><mailto:virbh...@gmail.
e options visit:
>>
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>>
>
>
> --
> _____
> -- Bandwidth and Colocation Provid
running with root user.
On Tue, Nov 29, 2011 at 1:52 AM, Edwin Lam wrote:
> 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:
> -
pport ?
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+91-8885268942
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s audio conference
> however it need resources (Processing + RAM) per additional line.
>
> ** **
>
> Regards,
>
> ** **
>
> Faisal Hanif
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digi
Thank you for sharing your exp. with me.
On Wed, Nov 30, 2011 at 7:34 PM, Darren Wiebe wrote:
> We've been happy with the polycom IP 7000.
>
> Darren Wiebe
> On Nov 30, 2011 1:40 AM, "virendra bhati" wrote:
>
>> Hi Faisal,
>>
>> Thanks for rep
n us for a live introductory 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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+91-88852
Hi All,
I read about the *Hint* in asterisk. I want to implements into my server
for testing purpose. How to use it ? please help me...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth
ANUNAVAIL'
-- Executing [h@bhati-test:1] NoOp("SIP/2218-02c3", "hangup the
call now") in new stack
haddock8-astrx*CLI> core show hint 2218
2218@bhati-subscribe : SIP/2218
State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the ri
;
> Regards,
> Sammy.
>
>
> On Tue, Dec 6, 2011 at 5:14 PM, virendra bhati wrote:
>
>> Hi All,
>>
>> I did some google and found some documents on that and finally I got some
>> response from asterisk . Below is the CLI output of my google.
>>
Hi All,
If you used *DEVICE_STATE *function then there is no need to used *HINT* it
work independently.
It's not become to confusion for me how to when to used *HINT *and
when *DEVICE_STATE
?
*
On Tue, Dec 6, 2011 at 6:20 PM, virendra bhati wrote:
> Hi All,
>
> Below bold app
Hi List,
Please tell me which ports should be required open for communication with
asterisk. like 5060 for sip calls, 4569 for IAX, 10,000 to 20,000..
Apart from these ports what else is required ?
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
port needs to be open as well. It also depends what
> other appliactions are running on asterisk-box which require port opening
> i.e apache or mysql etc.
>
> Regards,
> Sammy
>
>
> On Mon, Dec 12, 2011 at 3:21 PM, virendra bhati wrote:
>
>> Hi List,
>>
>
/mailman/listinfo/asterisk-users
>
>
>
> --
> _
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
>
will be appreciated
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Use awk to extract only the numeric value from output of above.
>
> Or you can use AMI to fetch sip peer details and parse the value you
> require.
>
>
> On Sun, Dec 18, 2011 at 10:26 AM, virendra bhati wrote:
>
>> Hi List,
>>
>> I have asterisk 1.6.2.20 install
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>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> asterisk
ached its number but the caller hung up
before the callee picked up.)") in new stack
-- Executing [1212@default:9] ExecIf("SIP/2209-0854",
"1?noop(Congestion. This status is usually a sign that the dialled number
is not recognised.)") in new stack
-- Executing [121
> 10.10.11.203" this is why you are getting congestion instead of NOANSWER.
> Fix that and add a timeout to your dial and it should work.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On B
.com --
> New to Asterisk? Join us for a live introductory 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 a
//www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
LI>
all information of application and function are missing but working without
an issue.
Is this problem due to asterisk upgrading. primarily asterisk was installed
with rpm (yum install asterisk) and later installed with Asterisk
1.6.2.20.tar.gz
--
Thanks and regards
Virendra Bhati
+91-88
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