[asterisk-users] AGI question
I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI question
in your dialplan: [context] ... h,1,AGI(...) David Ruggles [EMAIL PROTECTED] wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Have a burning question? Go to Yahoo! Answers and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI question
That's right, but i think that you should use: exten = h,1,DEADAGI( ) because in h extension the channel is considered as 'dead channel' , Regards, -- J. Espinal Slackware-es.com chester c young wrote: in your dialplan: [context] ... h,1,AGI(...) */David Ruggles [EMAIL PROTECTED]/* wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have a burning question? Go to Yahoo! Answers http://answers.yahoo.com/;_ylc=X3oDMTFvbGNhMGE3BF9TAzM5NjU0NTEwOARfcwMzOTY1NDUxMDMEc2VjA21haWxfdGFnbGluZQRzbGsDbWFpbF90YWcx and get answers from real people who know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agi Question
Hi everybody I'm working on an AGI in perl and i find this problem. My agi should dial an outside number (PSTN) but every time that i try to dial from the agi, asterisk shows this message: -- -- AGI Script Executing Application: (Dial) Options: (Zap/1d/95449078|30|tTrm) Aug 17 19:00:58 NOTICE[335890]: app_dial.c:654 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time -- my box it's working fine i'm using a E100P and i can dial to the pstn from any extension but when a try to do it from my agi i got this error and the call gets mude (no make any sound, finally goes to the timeout extension). i changed dialing a sip extension and it work fine, the code is -- print exec Dial Zap/1d/95449078|30|tTrm \n; $result = STDIN; -- So i don't know what to do, i'll be thankfull if you can help me with some ideas PD(i used the AGI perl package and i got the same problem) Thank you Att Yelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agi Question
On Tue, 2004-08-17 at 19:10, Yelson wrote: Hi everybody I'm working on an AGI in perl and i find this problem. My agi should dial an outside number (PSTN) but every time that i try to dial from the agi, asterisk shows this message: -- -- AGI Script Executing Application: (Dial) Options: (Zap/1d/95449078|30|tTrm) Aug 17 19:00:58 NOTICE[335890]: app_dial.c:654 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time Maybe it is me, but what is Zap/1d? For an E1, I don't think you should specify a channel but rather a group. So maybe the 1d should be g1. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI question
Hello, Can we see your dialplan related to that ? On Mon, 2004-05-03 at 13:40, Osvaldo Mundim wrote: Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question or something
Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does SET EXTENSION new extension allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database lookup from the script and then issue the SET EXTENSION command to asterisk to allocate the call to the right extension if available? I had to go to http://home.cogeco.ca /~camstuff/agi.html For the AGI command descriptions. Is there a better/more up to date site? Please any info would be great.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI question or something
yeah it works, we hadthe same requirementsin our agi applications:
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;my %input = $AGI-ReadParse();$AGI-setcallback(\mycallback);
$AGI-stream_file('beep');$AGI-set_extension('6');$AGI-set_priority(1);Azher
Nate Clifford [EMAIL PROTECTED] wrote:
Sorry for asking this question again butbefore I blow 100 dollars on a X100P I need to know this info:So does "SET EXTENSION " allow for you to set whichextension the rest of the call will occur over?So if a call comes into the switch and I could make the AGI script checkthe DID or DNIS which is really in the variable agi_dnid?After that I can do a database lookup from the script and then issue the"SET EXTENSION" command to asterisk to allocate the call to the rightextension if available?I had to go to http://home.cogeco.ca /~camstuff/agi.htmlFor the AGI command descriptions. Is there a better/more up to datesite?Please any info would be great..___Asterisk-Users mailing
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RE: [Asterisk-Users] AGI question or something
Do you have any good web references on AGI
or getting to that ACD type point with the switch?
Perl is no problem finding that ref material but Im new to the
Asterisk and AGI side of things.
Thanks for the response BTW.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin
Sent: Wednesday, October 29, 2003
8:10 PM
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] AGI
question or something
yeah it works, we hadthe same
requirementsin our agi applications:
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
$AGI-stream_file('beep');
$AGI-set_extension('6');
$AGI-set_priority(1);
Azher
Nate Clifford
[EMAIL PROTECTED] wrote:
Sorry for asking this question again but
before I blow 100 dollars on a X100P I need to know this info:
So does SET EXTENSION allow for you to set which
extension the rest of the call will occur over?
So if a call comes into the switch and I could make the AGI script check
the DID or DNIS which is really in the variable agi_dnid?
After that I can do a database lookup from the script and then issue the
SET EXTENSION command to asterisk to allocate the call to the right
extension if available?
I had to go to http://home.cogeco.ca /~camstuff/agi.html
For the AGI command descriptions. Is there a better/more up to date
site?
Please any info would be great..
___
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[EMAIL PROTECTED]
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Spears
RE: [Asterisk-Users] AGI question or something
A *bunch* of Asterisk resources, including AGI stuff is at
http://bugs.digium.com/bug_view_page.php?bug_id=434 If you know of
any resources not listed there, add them. I'm hoping Digium will put
these links on their web site under some kind of external or 3rd party
Asterisk resources page
On Wed, 2003-10-29 at 22:42, Nate Clifford wrote:
Do you have any good web references on AGI or getting to that ACD type
point with the switch?
Perl is no problem finding that ref material but Im new to the
Asterisk and AGI side of things.
Thanks for the response BTW.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin
Sent: Wednesday, October 29, 2003 8:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] AGI question or something
yeah it works, we had the same requirements in our agi applications:
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
$AGI-stream_file('beep');
$AGI-set_extension('6');
$AGI-set_priority(1);
Azher
Nate Clifford [EMAIL PROTECTED] wrote:
Sorry for asking this question again but
before I blow 100 dollars on a X100P I need to know this info:
So does SET EXTENSION allow for you to set which
extension the rest of the call will occur over?
So if a call comes into the switch and I could make the AGI
script check
the DID or DNIS which is really in the variable agi_dnid?
After that I can do a database lookup from the script and then
issue the
SET EXTENSION command to asterisk to allocate the call to
the right
extension if available?
I had to go to http://home.cogeco.ca /~camstuff/agi.html
For the AGI command descriptions. Is there a better/more up to
date
site?
Please any info would be great..
___
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Re: [Asterisk-Users] AGI question
On Sunday 14 September 2003 18:37, Emmkah wrote: Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of 0, but not hanging up either one? I found Cam Farnell's AGI documentation. Fine work, Cam, thanks a lot, but no clue in there. As far as I understand asterisk, the Dial application won't do this job either, would it? Well, two possibilities. First, check out the tT options to the Dial application, which allow a call to be interrupted without hanging up on either one (like to transfer to call parking). Second, check out the MeetMe conferencing application. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI question
Hi, sorry if this is a newbie question, but in fact I am sort of a newbie. Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of 0, but not hanging up either one? I found Cam Farnell's AGI documentation. Fine work, Cam, thanks a lot, but no clue in there. As far as I understand asterisk, the Dial application won't do this job either, would it? Any help is greatly appreciated. Rgds, Markus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
