Hi All
My dial plan has the following context:
[sip-guest]
exten = _!.,1, Answer
exten = _!.,n, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, resetcdr(w)
exten = _!.,n, resetcdr(w)
exten = _!.,n,
Hi All
Using Asterisk 11. My dial plan has the following context:
[sip-guest]
exten = _!.,1, Answer
exten = _!.,n, verbose(1,[${EXTEN}@${CONTEXT}])
exten = _!.,n, resetcdr(w)
exten = _!.,n, resetcdr(w)
exten = _!.,n, set(DNIS=${EXTEN})
exten = _!.,n, resetcdr(w)
exten = _!.,n,
To answer my question, set unanswered=yes in cdr.conf
Source:
http://lists.digium.com/pipermail/asterisk-users/2009-December/241749.ht
ml
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
I keep getting the following message whenever an AMI call is made:
asteriskjava.manager.internal.EventBuilderImpl.buildEvent(EventBuilderIm
pl.java:296) No event class registered for event type 'localbridge',
Tried adding an event listener. Anyone know how to fix this?
Thanks
Murthy
--
The LinkEvent is deprecated. Using asterisk-java-1.0.0.CI.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg
Sent: Thursday, October 31, 2013 1:27 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi All
I am unable to obtain CDR(answer) in AMI.
Tried the following:
$ telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
ActionID: 1
Username: admin
Secret: secret5
Action: Getvar
-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
On Wed, Oct 15, 2014 at 1:44 PM, Murthy Gandikota mgandik...@nts.net
wrote:
Hi All
I am unable to obtain CDR(answer) in AMI.
Tried the following:
$ telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost
I traced CDR(disposition) which was set to NO ANSWER. Apparently AMI
works the opposite of AGI in this case.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Wednesday, October 15, 2014 1
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Wednesday, October 15, 2014 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
I traced CDR(disposition) which was set to NO ANSWER
: Wednesday, October 15, 2014 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
On Wed, Oct 15, 2014 at 5:10 PM, Murthy Gandikota mgandik...@nts.net
wrote:
The CDR(disposition) is changing from context to context. Looks like
AGI
-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
On Wed, Oct 15, 2014 at 9:31 PM, Murthy Gandikota mgandik...@nts.net
wrote:
Thanks, Matthew. I think CDR(answer) is, in the end, not very useful to
me if it changes from context to context. Suppose from AMI we generate
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Jordan
Sent: Thursday, October 16, 2014 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
On Wed, Oct 15, 2014 at 9:31 PM, Murthy Gandikota mgandik
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Thursday, October 16, 2014 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
Apparently we are calling ResetCDR (not ForkCDR) in the Asterisk 11.5.1
-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Thursday, October 16, 2014 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AMI and CDR(answer)
Hi Matthew,
Now that you helped me figure out the root cause of my problem, I am
(answer)
On Thu, Oct 16, 2014 at 4:12 PM, Murthy Gandikota mgandik...@nts.net
wrote:
in cdr.c
void ast_cdr_reset(struct ast_cdr *cdr, struct ast_flags *_flags)
{
struct ast_cdr *duplicate;
struct ast_flags flags = { 0 };
if (_flags
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects there
are no channels in Asterisk - (sad) - but afterwards a PJSIP channel
from Alice enters into extension
12 Dialplan
On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota mgandik...@nts.net
wrote:
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects
] Asterisk 12 Dialplan
On Mon, Oct 27, 2014 at 10:56 AM, Murthy Gandikota mgandik...@nts.net
wrote:
Thanks, Richard. How do I get manager events such as VarSetEvent
(https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerEvent_Var
Set) using ARI
] Asterisk 12 Dialplan
On Mon, Oct 27, 2014 at 2:40 PM, Murthy Gandikota mgandik...@nts.net
wrote:
I am unable to detect the Manager_Setvar event using ARI.
Can you please let me know, in ARI lingo, the curl or javascript code to
detect the AMI Manager_Setvar event for myvar
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Monday, October 27, 2014 7:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan
From: asterisk-users-boun
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy
Gandikota
Sent: Tuesday, October 28, 2014 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 Dialplan
Tried this:
wscat -c
ws
Dialplan
On Wed, Oct 29, 2014 at 1:21 PM, Murthy Gandikota mgandik...@nts.net
wrote:
I am happy to report that
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Applications+REST
+API
has the answer to my dilemma. It seems an app has to subscribe to
channel
events before it can receive
If you have used sippy_cup by Ben Klang and Will Drexler please comment.
Please note, I know there is a Sipp users mailing list. I am trying to
catch the attention of the developers and users who work with asterisk as well.
I have a scenario where I expect field0 and field1 to be injected
to the
I would like to play audio--using controlplayback-- to 2 channels--agent and
caller- simultaneously. Tried meetme,confbridge,originate without success.
Tried redirecting the channels to a context, playing audio to the agent's
channel and then bridging the 2 channels. The problem with this is
that appkonference is an over-kill for an audio as its
main functionality is with video. Going past that.
Thanks
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota
Sent: Saturday, December 06, 2014 8:35 PM
To: Asterisk Users Mailing
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Murthy Gandikota
Sent: Monday, December 08, 2014 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Playing audio to bridged channels using
ControlPlayBack
There is one more
Hello All
What is the standard practice to adjust the volume on a channel? I am using App
Konference where they have a talk volume and listen volume. No matter what I
try, it's not making a difference. By the way, I know that the phone comes with
a volume control. I am interested in the
From: asterisk_l...@earthshod.co.uk
To: asterisk-users@lists.digium.com
Date: Mon, 3 Aug 2015 08:42:50 +0100
Subject: Re: [asterisk-users] Call Center
On Saturday 01 Aug 2015, Murthy Gandikota wrote:
Hi All
Has anyone used Asterisk for a Call Center operation? What I mean is: given
Date: Wed, 29 Jul 2015 11:47:19 -0500
From: sgriepent...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Windows Asterisk Help
On Wed, Jul 29, 2015 at 10:16 AM, John Novack jnov...@stromberg-carlson.org
wrote:
Murthy Gandikota wrote
Hi All
Has anyone used Asterisk for a Call Center operation? What I mean is: given a
list of phone numbers, can Asterisk dial each number, play a message and accept
some DTMF? I ask because I am an employee of a non-profit company based in San
Diego, CA. I already evaluated Voicent and Voxeo.
To: asterisk-users@lists.digium.com
From: webaccounts...@jgoettgens.de
Date: Wed, 29 Jul 2015 16:11:31 +0200
Subject: Re: [asterisk-users] Windows Asterisk Help
Downloaded latest version of Asterisk from
www.asteriskwin32.com and
Hi All,
Downloaded latest version of Asterisk from www.asteriskwin32.com and installed
on Windows 7.
Here is my sip.conf
[general]context = demo ; Default context for incoming
callsbindport = 5060 ; UDP Port to bind to (SIP standard port is
5060)bindaddr = 0.0.0.0
Aug 2015, Murthy Gandikota wrote:
For Asterisk INVITE please view
http://pastebin.com/v15vMax4
For X-Lite INVITE please view
http://pastebin.com/rmHZKu3N
Just a quick glance (because I'm not a SIP expert)...
(Asterisk)
Request-Line: INVITE sip:1732xxx@69.59.234.67:5061 SIP/2.0
(X
Date: Thu, 6 Aug 2015 12:37:36 -0700
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, 6 Aug 2015, Murthy Gandikota wrote:
[trimming cruft nobody cares about
Tested with X-Lite and it worked fiine. Is there some way to replace
Anonymous with a config parameter?
Thanks for your kind help
From: murth...@hotmail.com
To: asterisk-users@lists.digium.com
Subject: Asterisk uses Anonymous, but why?
Date: Wed, 5
...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota
murth...@hotmail.commailto:murth...@hotmail.com wrote:
Tested with X-Lite and it worked fiine. Is there some way to replace
Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudg...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota
murth...@hotmail.commailto:murth
Date: Thu, 6 Aug 2015 13:33:11 -0500
From: rmudg...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota
murth...@hotmail.commailto:murth
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota
murth...@hotmail.commailto:murth
Hi All
Noticed in sip.conf that the asterisk (v11) is sensitive to the order of peers.
Here is my sip.conf
[general]
context = demo ; Default context for incoming calls
bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr = 0.0.0.0 ;
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
a
href=http#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage
target=_blank
class=newlyinsertedlinkhttp#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage/a
It was not
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
a
href=http#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage
target=_blank
class=newlyinsertedlinkhttp#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage/a
It was not
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
a
href=http#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage
target=_blank
class=newlyinsertedlinkhttp#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage/a
It was not
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
a
href=http#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage
target=_blank
class=newlyinsertedlinkhttp#58;#47;#47;www.voip-info.org#47;wiki#47;view#47;Asterisk#43;and#43;Vonage/a
It was not
Hotmail barfed on me
--
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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:
Hi All
I am trying to do the following:
Set(msg=Hello ${world} how ${are} you)
I see that ${world} is substituted correctly but not ${are}
Using Asterisk 13
I am injecting ${world} and ${are} within an originate action (using
Asterisk-Java)
I understand one can use max 25 variables in a
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