Re: [Asterisk-Users] Meetme question
You need to either have a zap channel available or zap_dummy in place to get this working... SIP setup only requires a zap channel for meetme... K. Callis On Thu, 2003-09-25 at 00:43, C. Johnson wrote: > Ok.. I got * and SIP working internally now .. still wrestling with > connecting two remote * pbx's together.. I'll save that for another > day though :) > > I setup Meetme on this new * PBX, but when I try to dial to join the > conference, > I hear a recording saying the conference is invalid or something to > that effect. Here's a copy of my log files: > > == Parsing '/etc/asterisk/meetme.conf': Found > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to > open pseudo channel > > > It then hangs up.. Anyone seen this before?? > -cj > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme question
Have you got a zaptel device?? Can you post you meetne.conf? - Original Message - From: "C. Johnson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 3:43 PM Subject: [Asterisk-Users] Meetme question > Ok.. I got * and SIP working internally now .. still wrestling with > connecting two remote * pbx's together.. I'll save that for another > day though :) > > I setup Meetme on this new * PBX, but when I try to dial to join the > conference, > I hear a recording saying the conference is invalid or something to > that effect. Here's a copy of my log files: > > == Parsing '/etc/asterisk/meetme.conf': Found > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to > open pseudo channel > > > It then hangs up.. Anyone seen this before?? > -cj > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme question
> Ok.. I got * and SIP working internally now .. still wrestling with > connecting two remote * pbx's together.. I'll save that for another > day though :) > > I setup Meetme on this new * PBX, but when I try to dial to join the > conference, > I hear a recording saying the conference is invalid or something to > that effect. Here's a copy of my log files: > > == Parsing '/etc/asterisk/meetme.conf': Found > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to > open pseudo channel > > > It then hangs up.. Anyone seen this before?? > -cj 2 things to check.. 1) Do you have zaptel hardware in the Asterisk server? meetme requires at least one zaptel card or the use of ztdummy.. 2) Make sure you have a blank line at the bottom of the meetme.conf file after the last entry.. Strange I know but this threw me when I was trying to get meetme working.. :) Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
Basically what you need to do, unless you are installing zap devices, is uncomment the ztdummy line in the zaptel makefile, make install, and modprobe ztdummy. This will install a 'pseudo channel' driver for zap. (ie, emulating a zap device for applications needing zap interfaces). Hope this helps. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Johnson Sent: Thursday, September 25, 2003 1:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meetme question Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing '/etc/asterisk/meetme.conf': Found WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to open pseudo channel It then hangs up.. Anyone seen this before?? -cj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
Any idea how to setup a zap_dummy since I do not have a zap device?? Thanks :) -cj > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Kim C. Callis > Sent: Thursday, September 25, 2003 2:46 AM > To: Asterisk User Mailing List > Subject: Re: [Asterisk-Users] Meetme question > > > You need to either have a zap channel available or zap_dummy > in place to get this working... SIP setup only requires a zap > channel for meetme... > > K. Callis > > > > On Thu, 2003-09-25 at 00:43, C. Johnson wrote: > > Ok.. I got * and SIP working internally now .. still wrestling with > > connecting two remote * pbx's together.. I'll save that for another > > day though :) > > > > I setup Meetme on this new * PBX, but when I try to dial to > join the > > conference, I hear a recording saying the conference is invalid or > > something to that effect. Here's a copy of my log files: > > > > == Parsing '/etc/asterisk/meetme.conf': Found > > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable to > > open pseudo channel > > > > > > It then hangs up.. Anyone seen this before?? > > -cj > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
I was thinking faster than I type... That should have been ztdummy On Thu, 2003-09-25 at 01:02, C. Johnson wrote: > Any idea how to setup a zap_dummy since I do not have a zap device?? > > > Thanks :) > > -cj > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Kim C. Callis > > Sent: Thursday, September 25, 2003 2:46 AM > > To: Asterisk User Mailing List > > Subject: Re: [Asterisk-Users] Meetme question > > > > > > You need to either have a zap channel available or zap_dummy > > in place to get this working... SIP setup only requires a zap > > channel for meetme... > > > > K. Callis > > > > > > > > On Thu, 2003-09-25 at 00:43, C. Johnson wrote: > > > Ok.. I got * and SIP working internally now .. still wrestling > with > > > connecting two remote * pbx's together.. I'll save that for > another > > > day though :) > > > > > > I setup Meetme on this new * PBX, but when I try to dial to > > join the > > > conference, I hear a recording saying the conference is invalid or > > > > something to that effect. Here's a copy of my log files: > > > > > > == Parsing '/etc/asterisk/meetme.conf': Found > > > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable > to > > > open pseudo channel > > > > > > > > > It then hangs up.. Anyone seen this before?? > > > -cj > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/aster> isk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meetme question
No problem.. I am editing the makefile for zaptel now, and uncommenting ztdummy depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztdummy.o Hmm... Ok.. I give up.. I've been here for over 18 hours now, I'll try it with a fresh head tomorrow Thanks to all who replied (Kim, Chee, WipeOut, Josh...) -cj > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Kim C. Callis > Sent: Thursday, September 25, 2003 3:07 AM > To: Asterisk User Mailing List > Subject: RE: [Asterisk-Users] Meetme question > > > I was thinking faster than I type... That should have been ztdummy > > On Thu, 2003-09-25 at 01:02, C. Johnson wrote: > > Any idea how to setup a zap_dummy since I do not have a zap device?? > > > > > > Thanks :) > > > > -cj > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On Behalf Of > > > Kim C. Callis > > > Sent: Thursday, September 25, 2003 2:46 AM > > > To: Asterisk User Mailing List > > > Subject: Re: [Asterisk-Users] Meetme question > > > > > > > > > You need to either have a zap channel available or zap_dummy > > > in place to get this working... SIP setup only requires a zap > > > channel for meetme... > > > > > > K. Callis > > > > > > > > > > > > On Thu, 2003-09-25 at 00:43, C. Johnson wrote: > > > > Ok.. I got * and SIP working internally now .. still wrestling > > with > > > > connecting two remote * pbx's together.. I'll save that for > > another > > > > day though :) > > > > > > > > I setup Meetme on this new * PBX, but when I try to dial to > > > join the > > > > conference, I hear a recording saying the conference is > invalid or > > > > > > something to that effect. Here's a copy of my log files: > > > > > > > > == Parsing '/etc/asterisk/meetme.conf': Found > > > > WARNING[24592]: File app_meetme.c, Line 154 (build_conf): Unable > > to > > > > open pseudo channel > > > > > > > > > > > > It then hangs up.. Anyone seen this before?? > > > > -cj > > > > > > > > ___ > > > > Asterisk-Users mailing list [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/aster> isk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] meetme question
ï
Did
you uncomment the ztdummy in the zaptel Makefile?
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of christophe
de coninckSent: 21 October 2004 14:03To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] meetme
questionHi,I've just setup a meetme room like in the
config is set as demo, alltough when i try to call it I see
this: -- Executing MeetMe("SIP/christophe-9123",
"1234") in new stack == Parsing '/etc/asterisk/meetme.conf':
FoundOct 21 15:00:23 WARNING[409617]: chan_zap.c:755 zt_open: Unable to
open '/dev/zap/pseudo': No such file or directoryOct 21 15:00:23
ERROR[409617]: chan_zap.c:6663 chandup: Unable to dup channel: No such file or
directoryOct 21 15:00:23 WARNING[409617]: app_meetme.c:227 build_conf:
Unable to open pseudo channel - trying deviceOct 21 15:00:23
WARNING[409617]: app_meetme.c:230 build_conf: Unable to open pseudo
device -- Playing 'conf-invalid' (language
'en') == Spawn extension (default, 8600, 1) exited non-zero on
'SIP/christophe-9123'extensions.conf:exten =>
8600,1,Meetme(1234)exten =>
8600,2,HangUpmeetme.conf:[rooms]conf => 1234
-- Christophe De Coninck | Zarek K http://www.zarekk.bemailto: [EMAIL PROTECTED] mailto: [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
Re: [asterisk-users] Meetme question
Create two seperate extensions. One for the admin and one for the regular users that go to the same room. The issue you will have then is that the admin will have to call in first to create the dynamic room. - Original Message - From: "Adrian Marsh" <[EMAIL PROTECTED]> To: Sent: Saturday, March 31, 2007 3:58 PM Subject: [asterisk-users] Meetme question Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can join). What I really want is something that covers the below: - One call-in number - Employees get their own unique conference # (this could be their own extension), and can set a public PIN that only they can change. - I don't really want a www-based system, as most of my users are usually mobile, and might not have access to the corporate intranet. Thanks, Adrian Marsh ___ --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
Re: [asterisk-users] Meetme question
I have my system set up to check the cid of the calling number and if the room number the user inputs matches the calling extension (the last 4 digits in my case) then the number is considered admin. This does have the same downside that Dovid pointed out, the admin must be in the room for users to join. Dovid B wrote: Create two seperate extensions. One for the admin and one for the regular users that go to the same room. The issue you will have then is that the admin will have to call in first to create the dynamic room. - Original Message - From: "Adrian Marsh" <[EMAIL PROTECTED]> To: Sent: Saturday, March 31, 2007 3:58 PM Subject: [asterisk-users] Meetme question Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can join). What I really want is something that covers the below: - One call-in number - Employees get their own unique conference # (this could be their own extension), and can set a public PIN that only they can change. - I don't really want a www-based system, as most of my users are usually mobile, and might not have access to the corporate intranet. Thanks, Adrian Marsh ___ --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 ___ --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] Meetme question
Thanks Jason, Could you post some sample code? What do you do if the CLI is not present ? (i.e. international callee..) Thanks, Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Fuermann Sent: 02 April 2007 14:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Meetme question I have my system set up to check the cid of the calling number and if the room number the user inputs matches the calling extension (the last 4 digits in my case) then the number is considered admin. This does have the same downside that Dovid pointed out, the admin must be in the room for users to join. Dovid B wrote: > Create two seperate extensions. One for the admin and one for the > regular users that go to the same room. The issue you will have then > is that the admin will have to call in first to create the dynamic room. > > - Original Message - From: "Adrian Marsh" > <[EMAIL PROTECTED]> > To: > Sent: Saturday, March 31, 2007 3:58 PM > Subject: [asterisk-users] Meetme question > > > Hi, > > I'm experimenting with the Meetme feature of Asterisk 1.2, > > exten => 2095,1,MeetMe(|Ds) > > This almost gives me what I want, where each employee can create their > own on-the-fly conferences with a personal Conference Number and PIN. > However, as the PIN is actually set by the first callee, then its > subject to problems (first callee might enter the wrong PIN, and then > no-one else can join). > > What I really want is something that covers the below: > > - One call-in number > - Employees get their own unique conference # (this could be their own > extension), and can set a public PIN that only they can change. > - I don't really want a www-based system, as most of my users are > usually mobile, and might not have access to the corporate intranet. > > > Thanks, > > Adrian Marsh > > > ___ > --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 ___ --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 smime.p7s Description: S/MIME cryptographic signature ___ --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] Meetme question
How do I do that? Doesn't "Ds" create dynamic room numbers? Thanks, Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: 01 April 2007 11:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Meetme question Create two seperate extensions. One for the admin and one for the regular users that go to the same room. The issue you will have then is that the admin will have to call in first to create the dynamic room. - Original Message - From: "Adrian Marsh" <[EMAIL PROTECTED]> To: Sent: Saturday, March 31, 2007 3:58 PM Subject: [asterisk-users] Meetme question Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can join). What I really want is something that covers the below: - One call-in number - Employees get their own unique conference # (this could be their own extension), and can set a public PIN that only they can change. - I don't really want a www-based system, as most of my users are usually mobile, and might not have access to the corporate intranet. Thanks, Adrian Marsh ___ --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 ___ --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] Meetme Question
On Tue, September 13, 2005 11:53, Accursio Avona said: > Hi all, > I'd like to use the w option of the meetme application. > From tiki i read: > >'w' -- wait until the marked user enters the conference > > * All other connected users will hear MusicOnHold until the marked > user enters. > > The question is, how can i indicate the "marked user"? > Basically you create 2 different PINs for the meeting. One for normal users, one for the 'marked' user. It can then differentiate by comparing PINs. HTH -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Meetme Question
Accursio Avona wrote: Hi all, I'd like to use the w option of the meetme application. >From tiki i read: 'w' — wait until the marked user enters the conference * All other connected users will hear MusicOnHold until the marked user enters. The question is, how can i indicate the "marked user"? A quick search of the archives reveals: [EMAIL PROTECTED] wrote: On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote: Create an extension that the user to be marked knows about, maybe even have it authenticate, mark the user and drop them into the conference. Doug If the Marked user isn't the first to enter the channel, then how does the MeetMe app know to put all other users on hold until Marked user arrives? This is still unclear to me. Example: meetme.conf conf => 1000 extensions.conf ; ** Normal users enter the conference here ** exten => 4823,1,SetMusicOnHold(random) exten => 4823,2,Meetme(|Msciw) exten => 4823,3,Hangup() ; ** Extension to mark conference users* exten => 4824,1,Authenticate(12345) exten => 4824,2,Meetme(|Asci) exten => 4824,3,Hangup() Users using extension 4823 and entering conference 1000 will listen to hold music until the marked users enters. Users using extension 4824 and entering a password of 12345 will be able to select conference 1000 as the marked user. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Meetme Question
Hi, Thank you very much for your suggestion this was what i nedded. Best Regards Accursio Avona The question is, how can i indicate the "marked user"? A quick search of the archives reveals: Example: meetme.conf conf => 1000 extensions.conf ; ** Normal users enter the conference here ** exten => 4823,1,SetMusicOnHold(random) exten => 4823,2,Meetme(|Msciw) exten => 4823,3,Hangup() ; ** Extension to mark conference users* exten => 4824,1,Authenticate(12345) exten => 4824,2,Meetme(|Asci) exten => 4824,3,Hangup() Users using extension 4823 and entering conference 1000 will listen to hold music until the marked users enters. Users using extension 4824 and entering a password of 12345 will be able to select conference 1000 as the marked user. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Meetme Question
Hi , I am trying to add an operator assistance feature to meetme , when the user dials '0' ,support / help desk personnel should be added to the live conference for live support / troubleshooting. How can i do this ? Can I edit the meetme * menu and add a new menu item ' Press '0' for support' .I think I will have to edit the meetme.c source to do this , hard way :( or is it possible to write an AGI script which detects when a user dials '0' and calls the helpdesk number (preconfigured number) or generally is it possible to collect the DTMF response from a user during a meetme conf call and trigger some action / script , I searched a lot in forums / mailing list , most of the threads are pretty old and confusing. Any help / hints will be greatly appreciated. Thanks Shiju V.Joseph Just add "X" to the meetme string and define 0 action; something like this Exten => 1234,1,Goto(meetme-oper|s|1) [meetme-oper] Exten => s,1,meetme(1234,X) Exten => s,n,hangup Exten => 0,1,dial(SIP/100,30,m) When you dial 1234, you are put into conference 1234 If you press 0 while in the conference, you are transferred to extension 100. -- -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme Question
Hi Danny, Thanks a lot for the reply , I tried the dial plan you have provided , when pressing '0' it gets connected with the extension defined in [meetme-oper] context , but after disconnecting the operator call user exits from the meetme room , can we avoid that ? I am trying to achieve the following callflow, 1.User calls meetme bridge number 2.User enters the bridge with PIN 3.During the session if he needs any assistance he presses '0' and talks with the operator 4.After talking with the operator user gets back to the conference again Thanks in advance. Regards Shiju V.Joseph From: "Danny Nicholas" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Date: 07/21/2010 06:36 PM Subject: Re: [asterisk-users] Meetme Question Sent by: asterisk-users-boun...@lists.digium.com Hi , I am trying to add an operator assistance feature to meetme , when the user dials '0' ,support / help desk personnel should be added to the live conference for live support / troubleshooting. How can i do this ? Can I edit the meetme * menu and add a new menu item ' Press '0' for support' .I think I will have to edit the meetme.c source to do this , hard way :( or is it possible to write an AGI script which detects when a user dials '0' and calls the helpdesk number (preconfigured number) or generally is it possible to collect the DTMF response from a user during a meetme conf call and trigger some action / script , I searched a lot in forums / mailing list , most of the threads are pretty old and confusing. Any help / hints will be greatly appreciated. Thanks Shiju V.Joseph Just add ?X? to the meetme string and define 0 action; something like this Exten => 1234,1,Goto(meetme-oper|s|1) [meetme-oper] Exten => s,1,meetme(1234,X) Exten => s,n,hangup Exten => 0,1,dial(SIP/100,30,m) When you dial 1234, you are put into conference 1234 If you press 0 while in the conference, you are transferred to extension 100. _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The information contained in this communication is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. It may contain confidential or legally privileged information. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking any action in reliance on the contents of this information is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by responding to this email and then delete it from your system. Ernst & Young is neither liable for the proper and complete transmission of the information contained in this communication nor for any delay in its receipt. -- _ -- 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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
