I don't think so any such method to return variable from AGI. But simple
solution is set variable in AGI and then you can get back after AGI call in
dialplan and these variable will be available until call finished.
---
Virendra Bhati
+91-9718500594
+91-9250078532
Sr. Asterisk Developer
E-mail
Hi team,
I had implementation complete customized IPPBX solution with the help on
Asterisk , ARA and a2billing for billing purpose. Now only issue I come is
if a customer A and B want to used similar extension rang then it's only
possible with adding account-code like 100e12345 and 100e67890.
But
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
--
Virendra Bhati
+91-9718500594
+91-9250078532
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Virendra Bhati
+91-9718500594
+91-9250078532
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra
and regards
Virendra Bhati
+91-9718500594
+91-9250078532
Asterisk Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
Asterisk engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http
:
On Monday 21 October 2013, virendra bhati wrote:
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then
get
below issue.
*[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r
asterisk: error while loading shared libraries: libjansson.so.4: cannot
open
Thank you, My issue was resolved by provided information
On Thu, Oct 24, 2013 at 5:49 AM, Sylvain Boily sbo...@proformatique.comwrote:
Hello,
Le 2013-10-21 08:31, virendra bhati a écrit :
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then
get below issue
Yes I installed manually from tar file of jansson
On Wed, Oct 23, 2013 at 8:44 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Oct 21, 2013 at 7:26 AM, virendra bhati virbh...@gmail.comwrote:
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then
get below
, 2013 at 6:29 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Monday 21 October 2013, virendra bhati wrote:
Hi Team,
I have installed asterisk-12 Beta but when I try to asterisk start then
get
below issue.
*[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r
asterisk
-1-04 asterisk-12.0.0-beta1]#*
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
-1-04 asterisk-12.0.0-beta1]#*
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
--
_
-- Bandwidth and Colocation Provided by http://www.api
and regards
Virendra Bhati
+91-9718500594
+91-9250078532
Software Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id
: warning: incompatible implicit declaration of built-in
function âexitâ
make[1]: *** [eagi-test.o] Error 1
make: *** [agi-install] Error 2
[root@localhost asterisk-1.6.2.23]#
--
Thanks and regards
Virendra Bhati
+91-9250078532
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-9718500594
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[image: View my profile on
LinkedIn]http://in.linkedin.com/pub/virendra-bhati/6/a30/755
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
--
Thanks and regards
Virendra Bhati
+91-9718500594
, August 12, 2012 3:56 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] best free fax solution with asterisk
On 08/12/2012 10:32 AM, James Sharp wrote:
On 8/11/2012 8:05 AM, virendra bhati wrote:
Hi team,
I
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
--
Thanks and regards
Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail
if all other priority have 'n' as priority number?
In a relational database there is no 'sequential read'.
In other words, you need to assign the priority to all entries.
Leandro
Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha
scritto:
Hi Team,
I want to used *'n
Hi Team,
I want to used *'n*' as priority in asterisk realtime but asterisk don't
support n as next priority
I am using Asterisk 1.4.41
--
Thanks and regards
Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India
-26 at 23:20 +0530, virendra bhati wrote:
My sip.conf don't have any entry related to sip pees. I have
everything into database.
for more details please check below url, which have good example of
asterisk realtime
http://bahjons.com/stuff/asterisk-realtime-installation-guide
On Thu
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
--
Thanks and regards
Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype
]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE
If anyone have any suggestion please reply to me.
--
Thanks and regards
Virendra Bhati
+91
Of *virendra bhati
*Sent:* Thursday, July 26, 2012 10:35 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Asterisk Realtime issue after registering
withx-lite
Hi All,
I have an small issue, which is not creating any problem on working syatem
then correct me ...
--
Thanks and regards
Virendra Bhati
+91-9718300881
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
asterisk process and see if that user can touch files like that.
Regards,
Sammy
On Thu, Jul 5, 2012 at 10:47 PM, virendra bhati virbh...@gmail.comwrote:
Hi All,
It's small issue but making a big problem for my application. I have
CentOS release 5.8 (Final) with asterisk 1.4.41 installed. I
== Manager 'admin' logged off from 127.0.0.1
ks3098819*CLI
--
Thanks and regards
Virendra Bhati
+91-08885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hyderabad(India)
--
_
-- Bandwidth
...@gmail.com wrote:
Hi,
1- try putting absolute filepath in source and destination field.
2- verify that the permissions of the files you're changing.
Regards,
Sammy.
On Mon, May 21, 2012 at 5:10 PM, virendra bhati virbh...@gmail.comwrote:
Hi List,
I am trying to add new SIP account in new file
when you installed DAHDI/Zaptel on VM then it will work
On Mon, Mar 19, 2012 at 4:35 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
I am not sure whether my PRI / BRI card would detect in virtual machine. I
have to check.
On Sun, Mar 18, 2012 at 8:14 AM, virendra bhati virbh
://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hyderabad(India)
--
_
-- Bandwidth and Colocation
--
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
--
Thanks and regards
Virendra Bhati
+91
/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
--
_
-- Bandwidth
, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/22/2012 06:26 AM, virendra bhati wrote:
Does anyone know the correct information of my question. All are move
round and round .
What does that mean? I answered your question with the correct and
complete information.
On Tue, Feb 21, 2012 at 7
thanks for suggesting the link.
Yes i don't have networking, and good SIP communication knowledge.
On Wed, Feb 22, 2012 at 6:41 PM, Phil Frost p...@macprofessionals.comwrote:
On 02/22/2012 08:01 AM, virendra bhati wrote:
*Will these port of UDP, RPT [assume you mean RTP] or Both ?*
It's
Hi,
how many UDP ports is required for 1 call. and why .
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
--
_
-- Bandwidth and Colocation Provided by http
right now it's only voice call.
But thanks for segregate the call.
Now i want to know about all calls used port too.
On Tue, Feb 21, 2012 at 7:06 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/21/2012 07:30 AM, virendra bhati wrote:
Hi,
how many UDP ports is required for 1 call
ever get them to do it let me know ;)
-Bruce
On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes
steve-li...@geekinter.netwrote:
On 13 Feb 2012, at 12:06, virendra bhati wrote:
You can't set callerid for outgoing calls in case of PRI.
Why not? Every PRI I have used supported
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh
Hi List,
Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
technology FreeSwitch is used and asterisk don't. I don't know it's the
right or wrong but this question come to my mind...
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh
being that both are different in Architecture, Asterisk was
designed keeping PBX in mind but Freeswitch was for SIP switching
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
Why FreeSwitch can handle more then 1,000CC and asterisk only
as configuration and asterisk plan text file ?
FreeSwitch used sofia_sip and asterisk used sip ?
Asterisk is PBX and FreeSwitch is SoftSwitch ?
On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.fr wrote:
On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com
wrote:
Why FreeSwitch
@default:3] NoOp(Console/dsp,
**CONGESTION**) in new stack
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
--
_
-- Bandwidth and Colocation Provided by http
--
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
--
Thanks and regards
Virendra Bhati
+91
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
Hi,
Doing some changes on logger.conf and with the help of cli logger rotate
now problem is solved.
thank you Alec..
On Fri, Jan 27, 2012 at 2:35 PM, virendra bhati virbh...@gmail.com wrote:
Logger rotate is used to reload and start asterisk log of Events and
quesue.
And I want to store
how asterisk know that file name is changed ? why not asterisk make new
file with the name of *full* ?
Can someone please tell me this behaviour of Asterisk (1.6.2.20).
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
--
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
--
Thanks and regards
Virendra Bhati
+91
or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth
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 every Thurs:
http
cost while call is in progress.
One option that I was thinking is to check elapsed time by core show
channel channel-id and deduct the amount but we need to check it every
second or x seconds via AMI.
Regards,
Zohair Raza
On Wed, Jan 18, 2012 at 9:35 AM, virendra bhati virbh
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 every Thurs:
http
How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza
engineerzuhairr...@gmail.comwrote:
Hi,
Try setting CDR(clid)
Regards,
Zohair Raza
On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati virbh...@gmail.comwrote:
Hi,
I am using phpagi
/asterisk-users
--
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 every
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
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote:
Hi,
Give the complete details about the asterisk version, and SIP trunk conf
details
On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http
type as its Not convert request to video
Sent from my iPhone
On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, virendra bhati virbh...@gmail.com wrote:
Hi,
Please give you sip phone name and sip.conf and extensions.conf details
which is using for that communication.
And CLI output of asterisk is also
or something
I can manipulate ...
Sent from my iPhone
On ٠٣/٠١/٢٠١٢, at ٨:١٩ ص, virendra bhati virbh...@gmail.com wrote:
Which is means like if you are using sip 1234 then give the details of
[1234] into that open thread and relevent extensions details too
On Tue, Jan 3, 2012 at 11:30 AM
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 every Thurs:
http
()-SendDTMF(2)
SendDTMF(1)-- Read()
Put proper GOTOIFs after reads if you like.
--
Regards,
Sammy
On Thu, Dec 29, 2011 at 12:34 PM, virendra bhati virbh...@gmail.comwrote:
I originate calls from .call file and 1 channel I have at A server A and
another channel at B
()
same = n,NoOp(you are at help section)
same = n,Hangup()
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
LEG CALL --;
[senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds. Where n is length of IVR file in
seconds.
same = n,Wait(10)
same = n,SendDTMF(1)
--SATISH BAROT
On Wed, Dec 28, 2011 at 1:55 PM, virendra bhati
.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger pabelan...@digium.comwrote:
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets
Can you give an example how to set these oprion ...
On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.com wrote:
2011/12/27 virendra bhati virbh...@gmail.com
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except
in default context
named service
[default]
.
exten = service,1,NOOP(Incoming call from SIPp)
.
Regards,
Sammy
On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.comwrote:
Hi list,
I have installed SIPp into my server. But not able to used it properly.
how
section ?
On Tue, Dec 27, 2011 at 2:21 PM, Leandro Dardini ldard...@gmail.comwrote:
Yes, this is one of my entries:
[trunk1]
context=fromoutside
type=friend
deny=0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
2011/12/27 virendra bhati virbh...@gmail.com
Can you give an example how to set
'.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not found in
context 'default'.
haddock8-astrx*CLI
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
PM, Leandro Dardini ldard...@gmail.com wrote:
Yes, this is one of my entries:
[trunk1]
context=fromoutside
type=friend
deny=0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
2011/12/27 virendra bhati virbh...@gmail.com
Can you give an example how to set these oprion ...
On Tue, Dec 27
:
- Original Message -
Le 27/12/2011 16:04, Tim Nelson a écrit :
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati
virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my
handle_request_register:
Registration from '4411 sip:4411@204.152.194.246' failed for
'62.141.54.169' - Wrong password
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided
a databse with two fields: extension and password.
Then query the database with func_odbc function.
There is a spanish article about this:
http://www.voztovoice.org/?q=node/478
Regards
- Original Message -
*From:* virendra bhati virbh...@gmail.com
*To:* Asterisk Users Mailing List - Non
list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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
+91-8885268942
Software Engineer
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
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
Hi,
It will not work...
On Fri, Dec 23, 2011 at 3:18 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
So pipes can be used as a secondary delimiter?
On Fri, 2011-12-23 at 15:08 +0530, virendra bhati wrote:
Hi ,
make variable and then put in funtion GotoIf()
like
set(day=mon|wed|fri
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 every Thurs:
http
/mailman/listinfo/asterisk-users
--
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
:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
or answer the call and don't pick the call I always
get the same responce at asterisk.
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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 Behalf Of virendra bhati
Sent: Wednesday
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 virbh...@gmail.comwrote:
Hi List,
I have asterisk 1.6.2.20 installed at production server, I have 2 SIP
voip trunk for making outgoing and DID
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 every
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
--
Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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 virbh...@gmail.comwrote:
Hi List,
Please tell me which ports should
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
@bhati-subscribe : SIP/2218
State:Idle
Watchers 0
1 hint matching extension 2218
*
*Is this the right way to use HINT of asterisk ?*
On Tue, Dec 6, 2011 at 3:38 PM, virendra bhati virbh...@gmail.com wrote:
Hi All,
I read about the *Hint* in asterisk. I want to implements into my server
, virendra bhati virbh...@gmail.com 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.
*haddock8-astrx*CLI core show hint 2218
2218@bhati-subscribe : SIP/2218
State:Idle
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 virbh...@gmail.com wrote:
Hi All,
Below bold
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 every Thurs:
http://www.asterisk.org
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.digium.com] *On Behalf Of *virendra bhati
*Sent:* Wednesday, November 30
Thank you for sharing your exp. with me.
On Wed, Nov 30, 2011 at 7:34 PM, Darren Wiebe dar...@aleph-com.net wrote:
We've been happy with the polycom IP 7000.
Darren Wiebe
On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote:
Hi Faisal,
Thanks for reply but I want hardware
?
--
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 every Thurs
1 - 100 of 229 matches
Mail list logo
