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
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 i
ttp://www.api-digital.com --
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Skype id:- virbhati2
New Delhi(India)
[ima
>
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>
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+91-9718500594
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E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
[im
D section. earlier is was *[CallerIDnum]
*
*So 'n' is now 'N'
*
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+91-9718500594
+91-9250078532
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Skype id:- virbhati2
New Delhi(India)
[image
Thank you, My issue was resolved by provided information
On Thu, Oct 24, 2013 at 5:49 AM, Sylvain Boily wrote:
> 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 be
Hi Team,
After suggested links and patch ,
I installed all and then start asterisk and that start working.
Thanks for suggestion..
On Wed, Oct 23, 2013 at 3:55 AM, Matthew Jordan wrote:
>
> On Mon, Oct 21, 2013 at 7:59 AM, A J Stiles > wrote:
>
>> On Monday 21 October 20
o.1 => /lib64/libselinux.so.1 (0x00345800)*
[root@cs-gb-pwr-1-04 ~]#
On Mon, Oct 21, 2013 at 6:29 PM, A J Stiles
wrote:
> On Monday 21 October 2013, virendra bhati wrote:
> > Hi Team,
> >
> > I have installed asterisk-12 Beta but when I try to asterisk start t
Yes I installed manually from tar file of jansson
On Wed, Oct 23, 2013 at 8:44 AM, Warren Selby wrote:
> On Mon, Oct 21, 2013 at 7:26 AM, virendra bhati wrote:
>
>> Hi Team,
>>
>> I have installed asterisk-12 Beta but when I try to asterisk start then
>> get below
-1-04 asterisk-12.0.0-beta1]#*
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+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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-1-04 asterisk-12.0.0-beta1]#*
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Virendra Bhati
+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/
hello
>
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> To UNSUBSCRIBE or update options visit:
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; http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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+91-9718500594
+91-9250078532
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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llo
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E-m
options visit:
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+91-9718500594
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E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New
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/
âexitâ
eagi-test.c:168: 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]#
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Virendra Bhati
+91-9250078532
Asterisk Developer
E-mail-: virbh...@gmail.c
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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Vire
ling list
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+91-9718500594
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
; 616-855-1030 Ext. 2003
>>
>>
>> --
>> *From*: "Steve Underwood"
>> *Sent*: Sunday, August 12, 2012 3:56 AM
>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
>> asterisk-users@list
lman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>>>
>>
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk?
priority 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" ha
> s
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
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+91-9718300881
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E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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.c
k-users-boun...@lists.digium.com] *On Behalf 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 h
is not a
valid host
[Jul 26 21:20:37] NOTICE[17811]: 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 pleas
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nning 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 wrote:
>
>> Hi All,
>>
>> It's small issue but making a big problem for my application. I have
>> CentOS release 5.8 (F
If I am wrong then correct me ...
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Virendra Bhati
+91-9718300881
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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I have update sammy but no luck
On Mon, May 21, 2012 at 5:49 PM, SamyGo 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,
*CLI Log:-*
ks3098819*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
ks3098819*CLI>
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Virendra Bhati
+91-08885268942
Software Engineer
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:
lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>>
>> Dog is my Co-pilot
>>
>>
>
> --
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> New to
gt; To UNSUBSCRIBE or update options visit:
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Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing l
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 wrote:
> On 02/22/2012 08:01 AM, virendra bhati wrote:
>
>> *Will these port of UDP, RPT [assume you mean RTP] or Both ?*
>>
&
, Kevin P. Fleming wrote:
> 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.
&g
isk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
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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 wrote:
> On 02/21/2012 07:30 AM, virendra bhati wrote:
>
>>
>> Hi,
>>
>> how many UDP po
Hi,
how many UDP ports is required for 1 call. and why .
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+91-8885268942
Software Engineer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
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If you ever get them to do it let me know ;)
>>
>> -Bruce
>>
>>
>> On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes
>> wrote:
>>
>>> On 13 Feb 2012, at 12:06, virendra bhati wrote:
>>> > You can't set callerid for outgoing calls in case
> musiconhold=default
>> immediate=no
>> txgain=0.0
>> rxgain=0.0
>> overlapdial=yes
>>
>>
>>
>>
>>
>
>
> --
> Unai Solaguren.
> usolagu...@gmail.com
>
> --
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ML 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 wrote:
> On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati
> wrote:
> >Why FreeSwitch can handle more
eing 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 wrote:
>
>> Hi List,
>>
>> Why FreeSwitch can handle m
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...
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uction of SIP dialog '
3cd12da658b42c10186c01ed3a7d2...@sip.voipon.co.uk' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [00918885268942@default:3] NoOp("Console/dsp",
"**CONGESTION**") in new stack
--
Th
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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 wrote:
> Logger rotate is used to reload and start asterisk log of Events and
> quesue.
>
> And I want to sto
erisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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.
So 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).
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E-mail-: virbh...@gmail.com
Skype id:- virb
t
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Kevin,
Dialogic doesn't provide any soultion as open source. It provides hardware
base cards for making outbond calls. And they used asterisk as backend for
they card application.
On Thu, Jan 19, 2012 at 6:50 PM, Kevin P. Fleming wrote:
> On 01/18/2012 11:14 PM, virendra bhati wrote:
gt; http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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> asterisk-users ma
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, 201
ing list
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How to used it in AGI ? I think it's Dialplan apps.
On Thu, Jan 12, 2012 at 2:18 PM, Zohair Raza
wrote:
> Hi,
>
> Try setting CDR(clid)
>
> Regards,
> Zohair Raza
>
>
>
>
>
> On Thu, Jan 12, 2012 at 12:44 PM, virendra bhati wrote:
>
>> Hi,
&
v-sip-trunk-out-f1.gradwell.net, port 5060
-- Called 00918885268...@sip.trunk.gradwell.com
[Jan 12 14:10:52] WARNING[28001]: chan_sip.c:18463 handle_response_invite:
Received response: "Forbidden" from '"01133200274" <
sip:01133200274@10.10.10.181>;tag=as76229e
gital.com --
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i got F=sip:test02@anonymous.invalid in sip header. I dont
> know why asterisk sends anonymous.invalid instead of domain name..Help me
>
>
> Best Regards,
> *Jayesh Labade*
> e-mail: jayesh.lab...@gmail.com
>
>
>
> On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati wro
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&
me.
[See Also]
Not available
On Tue, Jan 3, 2012 at 11:53 AM, Faraj Khasib wrote:
> Here is the thing, my sip client can call the same. Extension once as
> audio and once as video, so I cannt turn off video supportat reciever, what
> I guess can be done is in extension.conf , there must be fla
.. Not convert request to video
>
> Sent from my iPhone
>
> On ٠٣/٠١/٢٠١٢, at ٧:٢٩ ص, "virendra bhati" 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
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--
_
--
dDTMF(3)--> Read()
> Read()<-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 wrote:
>
>> I or
ork on request and responce
then it will be the last solution as per my knowledge.
Is this possible with the dialplan or I am just westing time?
On Wed, Dec 28, 2011 at 10:29 PM, Paul Belanger wrote:
> On 11-12-28 03:25 AM, virendra bhati wrote:
>
>> Hi list,
>>
>> Is ther
tleast '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 wrote:
>
>> Hi list,
>>
>> Is there any w
,Hangup()
exten=> help,1,Answer()
same => n,NoOp(you are at help section)
same => n,Hangup()
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age -
> > 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:
> > >>
> > >>
> > &g
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 wrote:
>
>> Hi list,
>>
>> I have installed SIPp int
default'.
== 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'.
== Using SIP RTP CoS mark 5
[Dec 27 17:37:55] NOTICE
Thank you Leandro,
Now i am able to register with fix IP.
On Tue, Dec 27, 2011 at 3:10 PM, Leandro Dardini wrote:
> With deny you'll "deny" all IP
> with permit you'll "permit" only your IP.
>
> Yes, it is mandatory to define both deny and permit.
&
1 at 2:21 PM, Leandro Dardini 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
>
>> Can you give an example how to set these oprion
Can you give an example how to set these oprion ...
On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini wrote:
>
>
> 2011/12/27 virendra bhati
>
>> Hi list someone is trying to hack my server . Is there any way by whcih I
>> can stop hacking of my server except iptables
9' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '"4411" ' failed for
'62.141.54.169' - Wrong password
[Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register:
Registration from '&qu
eate 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
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Hi,
It will not work...
On Fri, Dec 23, 2011 at 3:18 PM, Ishfaq Malik 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()
> >
y http://www.api-digital.com --
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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
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Virendra Bhati
+91-88
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Thanks a
> 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
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
_
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>> asterisk
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
will be appreciated
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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>
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>
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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,
>>
>
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 ?
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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
;
> 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.
>>
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
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...
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
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n us for a live introductory webinar every Thurs:
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