[asterisk-users] MeetMe and dynamic_features
Hello, I am trying to use a dynamic_features during a MeetMe conference without any luck. The dynamic_features defined macro works great during a normal call, but is ignored while on a MeetMe conference. extensions.conf [macro-RaiseHand] exten => s,1,DumpChan(1) features.conf RaiseHand => #5,peer/caller,Macro(RaiseHand) extensions.ael Set(DYNAMIC_FEATURES=RaiseHand); MeetMe(1234,F); I have tried with and without the F parameter... Any suggestion? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme vs confbridge max user comparison wanted
On Mon, Apr 13, 2015 at 1:15 PM, Steve Edwards asterisk@sedwards.com wrote: I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme and I'd like to switch to confbridge to service more callers. Can anyone reply with their experience along the lines of 'using meetme I was only getting x callers per server but with confbridge I now get y callers per server?' Anecdotally, when ConfBridge was first rewritten in Asterisk 10, some performance comparisons with MeetMe were performed. In the best case, on a particular system with conference user usage patterns, we saw MeetMe hit a limit at around 60 channels, and ConfBridge reach over 240 channels. Worst case for ConfBridge was around 140 channels. Note that the ConfBridge sample rate, mixing interval, and other parameters can greatly affect how far it scales out. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] meetme vs confbridge max user comparison wanted
I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme and I'd like to switch to confbridge to service more callers. Can anyone reply with their experience along the lines of 'using meetme I was only getting x callers per server but with confbridge I now get y callers per server?' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
[asterisk-users] MeetMe - Howto put in talk only mode using CLI/AMI
Hi, is there a way to put a conference participant in talk only mode (not listening) using CLI or AMI like mute/unmute ? MeetMe in Asterisk 1.8 Thanks for any hint. -- Daniel -- _ -- 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
[asterisk-users] MeetMe conference splitting
Hello, How to move 2 of 3 users in the MeetMe conference to the newly created MeetMe conference? Dialplan example is welcome. Best, Igor -- _ -- 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 conference splitting
On Thu, Jan 23, 2014 at 8:09 AM, Igor Dvorzhak idm...@gmail.com wrote: snip How to move 2 of 3 users in the MeetMe conference to the newly created MeetMe conference? Dialplan example is welcome. Maybe something like an AMI redirect? https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Redirect https://wiki.asterisk.org/wiki/display/AST/AMI+Examples -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] Meetme Show Activity in Minus
Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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 Show Activity in Minus
Solved On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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 conference password and time limitation
So this web-meetme applicationrequires to enable the real time in asterisk? Where I can find documentation about web-meetme application? Regards Bilal On Tuesday, October 1, 2013 6:57 PM, Dan Austin dan_aus...@phoenix.com wrote: Look at Web-MeetMe ( http://sf.net/projects/web-meetme ) If you are on Asterisk 1.6.7 or later you have access to RealTime MeetMe conference storage, otherwise you need to use a script and Asterisk application included with the WMM download. Dan From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 01, 2013 12:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme conference password and time limitation Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time .. Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to read/write from database and do the meetme conference within the AGI script it self, or there is simpler method? Regards Bilal-- _ -- 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
[asterisk-users] meetme conference password and time limitation
Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time .. Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to read/write from database and do the meetme conference within the AGI script it self, or there is simpler method? Regards Bilal-- _ -- 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 conference password and time limitation
Look at Web-MeetMe ( http://sf.net/projects/web-meetme ) If you are on Asterisk 1.6.7 or later you have access to RealTime MeetMe conference storage, otherwise you need to use a script and Asterisk application included with the WMM download. Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 01, 2013 12:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme conference password and time limitation Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time .. Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to read/write from database and do the meetme conference within the AGI script it self, or there is simpler method? Regards Bilal -- _ -- 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 and setting conference timeout
exten = 123,1,Set(TIMEOUT(absolute)=3600) exten = 123,n,MeetMe(blah,d) if you are using freepbx and you want to set timeout for all conference rooms go here -http://dn.forceit.ru/asterisk-conference-timeout -- _ -- 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
[asterisk-users] MeetMe Admin unmute user problem
Hello fellow asterisk users, I've been facing a problem when using MeetMe's admin functionality to unmute users in a conference using *Asterisk 1.6.2.11*. I've tried: 1) MeetMeUnmute (AMI) 2) MeetMeAdmin(AMI) 3) MeetMeChannelAdmin(AMI) and also tried via console : asterisk -rx 'meetme unmute conf_no user_no' and the available AGI functions. but all of this to no avail. The only output error debug that I get in the logs when a admin presses unmute key to unmute a user is: * * *MEETMEADMINSTATUS= NOTFOUND* * * Though users can mute/unumute themselves fine by pressing the mute/unmute key. Another thing that I've noted is that if I enter the above mentioned commands via a telnet session to the asterisk server they work fine. Any help or input will be appreciated. Millhouse * * -- _ -- 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 list concise
Sent from my Verizon Wireless 4G LTE DROID Dan Austin dan_aus...@phoenix.com wrote: -- _ -- 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-- _ -- 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
[asterisk-users] meetme list concise
Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- 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 and dtmf
I don't get what the 'F' option is for. Its not proper to exit a context and then reenter the conference as admin Isn't there any other way to do actions such as kick/mute/unmute users by admin dtmf trigger? On Fri, Jun 1, 2012 at 3:47 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 31 May 2012, Daniel Knoll wrote: is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. I'm just a 1.2 Luddite, but... You can use the meetme() 'X' option to jump out of the meetme and into another context. I use this to allow conference administrators to mute, un-mute, or kick users. The first digit jumps out of the meetme and into another context where I read additional digits (the user index) and then call an AGI (meetmeadmin-by-index) before returning the admin to the conference. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list concise
This list was accurate up to and including Asterisk 11 [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] meetme list concise Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- 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 list concise
Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x There doesn't seem to be any interface for [8] = Requests Floor. How can we put initially muted users in the request to talk queue? The provision of this parameter in the meet-me source indicates this is doable... but I am unable to find an appropriate way to do it. Any hints would be great help. On Thu, Aug 15, 2013 at 11:03 PM, Dan Austin dan_aus...@phoenix.com wrote: This list was accurate up to and including Asterisk 11 ** ** [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration ** ** Dan ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ® *Sent:* Thursday, August 15, 2013 4:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] meetme list concise ** ** Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- 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 -- _ -- 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 list concise
The only way that I know of, and it may not be in all of the 1.6 series, is to use the telephone menu (*5) I think, but would need to dig through the code. Dan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme list concise Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x There doesn't seem to be any interface for [8] = Requests Floor. How can we put initially muted users in the request to talk queue? The provision of this parameter in the meet-me source indicates this is doable... but I am unable to find an appropriate way to do it. Any hints would be great help. On Thu, Aug 15, 2013 at 11:03 PM, Dan Austin dan_aus...@phoenix.commailto:dan_aus...@phoenix.com wrote: This list was accurate up to and including Asterisk 11 [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration Dan From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of [Digital^Dude] (r) Sent: Thursday, August 15, 2013 4:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] meetme list concise Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) Thanks -- _ -- 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 -- _ -- 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 and maxusers option
Thiago wrote: I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Asterisk version? Any error messages? Is the conference you are attempting to limit stored in a db (Realtime)? Dan -- _ -- 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 and maxusers option
On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote: 2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. 2013-07-22 16:59, Thiago Coutinho skrev: Hi Johan. But the option maxusers should work too, right? I guess so, but I have not used it myself. It's not very hard to build you own dialplan with func_odbc and custom tables. This way you could use Meetme, Confbridge, or something else to do the mixing. -- Johan Wilfer -- _ -- 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 and maxusers option
Hi Johan. But the option maxusers should work too, right? On Fri, Jul 19, 2013 at 2:52 PM, Johan Wilfer li...@jttech.se wrote: 2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. -- Johan Wilfer -- _ -- 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 -- thiagoc O povo não deveria temer o governo. O governo é quem deveria temer o povo. V de Vingança -- _ -- 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
[asterisk-users] Meetme and maxusers option
Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? -- thiagoc O povo não deveria temer o governo. O governo é quem deveria temer o povo. V de Vingança -- _ -- 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 and maxusers option
2013-07-19 15:35, Thiago Coutinho skrev: Hi all. I'm trying to limit the number of participants in a conference room with the realtime option maxusers, but it doesn't work. Someone have this option working properly? Try these: https://wiki.asterisk.org/wiki/display/AST/Function_GROUP https://wiki.asterisk.org/wiki/display/AST/Function_GROUP_COUNT This is how I do it. This way you can do it more flexible in the dialplan. -- Johan Wilfer -- _ -- 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
[asterisk-users] meetme configuration
hello list , i want to use meetme with asterisk1.4 i check in this forum and i found this code : exten = 508,1,MeetMe(1000,ipdM) when i use this code in my server i can say my name and i press 1 in order to enter in the conference ; but i want to asks the customer to press an number and password in order to join this conference could you please give me an example thanks and regards -- _ -- 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 exit status?
On 06/02/2013 08:36 PM, Patrick Lists wrote: Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick There is no channel variable that provides that level of granularity. The closest available is the MEETMESECS channel variable, which tells you how many seconds the participant was in the conference. You can find a full list on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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 exit status?
On 06/03/2013 06:47 PM, Matthew Jordan wrote: On 06/02/2013 08:36 PM, Patrick Lists wrote: Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick There is no channel variable that provides that level of granularity. The closest available is the MEETMESECS channel variable, which tells you how many seconds the participant was in the conference. You can find a full list on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables Thanks Matt. I'll see if I can use MEETMESECS. Regards, Patrick -- _ -- 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
[asterisk-users] MeetMe exit status?
Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello, what is the equivalent parameter of X in the ConfBridge()-command ? How can you exit ConfBridge by pressing a digit ? Concerning MeetMe() : Verbosity is 25 and I still don't see anything on the console or in the logs when pressing '0' (zero). Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellensjonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. Please don't top post (https://www.asterisk.org/community/discuss). Also, you didn't pastebin any debug, so I can't confirm that there is not some other issue upon a possible DTMF reception. If it is the case that Asterisk doesn't detect a DTMF 0 when you send it from the endpoint, then you probably want to look at a SIP packet capture to verify the endpoint is actually sending the DTMF to Asterisk. What you look for in the capture or audio will depend on what kind of DTMF you are sending with the endpoint. Does Asterisk detect the digit 0 at any other time outside of MeetMe? Can you setup an extension matching for 1234567890 and dial that? Do you see DTMF debug for all those digits? If you do end up trying ConfBridge - I've never used it in 1.8. Others have made me aware that ConfBridge wasn't the best in 1.8, and that it's much better in 10 or preferably 11. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 Hello, I've tried now from Cisco SPA 508G and from Yealink T-28 to exit Meetme() by pressing '0' (zero) but no success. As I said, to log in I need to give password 12340 and that goes very well ! Once inside the conference room, I can press any digit : nothing happens. Nothing in the logs about DTMF being received. To exit the whole conferencing thing I can press # and that also succeeds ! So I don't think it has anything to do with DTMF-troubles. I've taken a pcap trace on the Yealink T-28. Where can I find the DTMF ? I can filter SIP, but no DTMF. To check if they were well send... Kind regards, Jonas. -- _ -- 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 and MEETME_EXIT_CONTEXT
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be I've tried now from Cisco SPA 508G and from Yealink T-28 to exit Meetme() by pressing '0' (zero) but no success. As I said, to log in I need to give password 12340 and that goes very well ! Once inside the conference room, I can press any digit : nothing happens. Nothing in the logs about DTMF being received. To exit the whole conferencing thing I can press # and that also succeeds ! So I don't think it has anything to do with DTMF-troubles. I've taken a pcap trace on the Yealink T-28. Where can I find the DTMF ? I can filter SIP, but no DTMF. To check if they were well send... If DTMF including 0 and all of the digits work fine most places in Asterisk other than MeetMe, then it looks like you are having a problem with the MeetMe application specifically. Especially considering that you have tried two different phones (which *should* rule out a phone as the issue). I attempted to reproduce your problem by following your dialplan and it worked fine for me in the latest SVN version of 1.8. Can you try upgrading to the 1.8.20 and see if the problem goes away? Maybe there was an issue that was fixed.. If the problem occurs in the latest release of 1.8.X then you may want to file a bug on the issue tracker. Please read through https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines before filing an issue, and be sure to include an Asterisk full log with VERBOSE and DEBUG messages captured at the same time as a SIP packet capture including all RTP as well. Let us know what happens.. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- 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
[asterisk-users] Meetme and MEETME_EXIT_CONTEXT
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten = _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten = _1000X,n,MeetMe(${CONFNO},dMX)//
//
//
//[dynamic-nway-invite]//
//exten = 0,1,NoOp(confno = ${CONFNO})//
//exten = 0,n,Read(DEST,dial,,i)//
//exten = 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)//
//exten = 0,n,Dial(Local/${DEST}@${LocalContext},,g)//
//exten = 0,n,Set(DYNAMIC_FEATURES=)//
//exten = 0,n,NoOp(tralalala)//
//exten = 0,n,Goto(dynamic-nway1,${CONFNO},1)//
//exten = i,1,Goto(dynamic-nway1,${CONFNO},1)//
/
So by pressing 0 (zero) while in the conference room, I should be able
to exit and continue in the context [dynamic-nway-invite] . Correct ?
But nothing happens when pressing 0 (zero).
What am I missing ??
Kind regards,
Jonas.
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Re: [asterisk-users] Meetme and MEETME_EXIT_CONTEXT
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello, I don't really see anything when pressing '0' (zero). It's like the '0' (zero) does not reach Asterisk. However the password to enter the conference does reach Asterisk well. Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- 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 and MEETME_EXIT_CONTEXT
- Original Message - From: Jonas Kellens jonas.kell...@telenet.be Hello, I don't really see anything when pressing '0' (zero). It's like the '0' (zero) does not reach Asterisk. However the password to enter the conference does reach Asterisk well. Please don't top post (https://www.asterisk.org/community/discuss). Also, you didn't pastebin any debug, so I can't confirm that there is not some other issue upon a possible DTMF reception. If it is the case that Asterisk doesn't detect a DTMF 0 when you send it from the endpoint, then you probably want to look at a SIP packet capture to verify the endpoint is actually sending the DTMF to Asterisk. What you look for in the capture or audio will depend on what kind of DTMF you are sending with the endpoint. Does Asterisk detect the digit 0 at any other time outside of MeetMe? Can you setup an extension matching for 1234567890 and dial that? Do you see DTMF debug for all those digits? If you do end up trying ConfBridge - I've never used it in 1.8. Others have made me aware that ConfBridge wasn't the best in 1.8, and that it's much better in 10 or preferably 11. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello, what is the equivalent parameter of X in the ConfBridge()-command ? How can you exit ConfBridge by pressing a digit ? Concerning MeetMe() : Verbosity is 25 and I still don't see anything on the console or in the logs when pressing '0' (zero). Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- 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 on short network
Jerry Geis wrote: I am running asterisk 1.4.43 on a really small network for testing, all on same switch. I launch a meetme between my server and 5 asterisk clients that are all on 10 foot network cables all connected to the same switch. The meetme is fine everything is in sync Then I reboot one of the clients. When it reboots I automatcially bring it back into the conference. however now its not really in sync. By not in sync do you mean that there is a delay between when the speaker speaks and when the client hears it? I'm trying to understand why that might be??? I thought it would. The conference is a listen only conference. Its not off or out of sync by much - but it is noticable. There's always going to be some amount of delay. It takes time to encode the audio, send it, mix it (in this case), receive it, decode it, and have it pass through a jitterbuffer (which by definition of being a buffer introduces delay). How much of a delay are you hearing? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 on short network
By not in sync do you mean that there is a delay between when the speaker speaks and when the client hears it? There's always going to be some amount of delay. It takes time to encode the audio, send it, mix it (in this case), receive it, decode it, and have it pass through a jitterbuffer (which by definition of being a buffer introduces delay). How much of a delay are you hearing? Josh, I am using a source file so its not speak live voice. when I say not in sync I don't care about a delay per say - its that 4 out of 5 of the clients are saying the same thing at the same time and the one I rebooted is just slightly off not identical to 1-4. So I have 5 clients where the hardware is identical, on the same switch, same length network cable, etc... I reboot one unit so even though it goes away and then rejoins the MeetMe - I would think that the server is still sending out audio at the same time as the other 1-4 units and would take the exact same amount of time to decode and all that and should be in sync with clients 1-4. again - dont care about delay - was just expecting the 1-5 units to all be in sync. Thanks Jerry -- _ -- 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
[asterisk-users] Meetme on short network
I am running asterisk 1.4.43 on a really small network for testing, all on same switch. I launch a meetme between my server and 5 asterisk clients that are all on 10 foot network cables all connected to the same switch. The meetme is fine everything is in sync Then I reboot one of the clients. When it reboots I automatcially bring it back into the conference. however now its not really in sync. I'm trying to understand why that might be??? I thought it would. The conference is a listen only conference. Its not off or out of sync by much - but it is noticable. Is there anything I can do to make that audio more in sync all the time? Thanks, Jerry -- _ -- 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 race condition
Jerry Geis wrote: I think I have a race condition. I am running something like this in my dialplan call agi to bring my list of devices into my MeetMe Playback beep start MeetMe() So in fact the meetme is not started before I bring the list of devices into the meetme. How can I do this differently so the MeetMe is started first or how can I wait in my AGI on the MeetMe to start because the MeetMe wont start until I exit the AGI... - or how do I in the dialplan wait for for the Meetme because I do have a stage where I redirect the Call into the MeetMe. so how do I inject a line that waits there for the MeetMe to be active??? Can you clarify what you mean by MeetMe to be active? What MeetMe options are you using and what is your configuration like? With the proper combination of options it shouldn't matter who gets into the conference bridge first. This is what Page essentially does, with the difference being that only the channel executing Page() can talk. If that behavior is what you are trying to accomplish I suggest you use that instead. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 race condition
Can you clarify what you mean by MeetMe to be active? What MeetMe options are you using and what is your configuration like? With the proper combination of options it shouldn't matter who gets into the conference bridge first. This is what Page essentially does, with the difference being that only the channel executing Page() can talk. If that behavior is what you are trying to accomplish I suggest you use that instead. Josh Well I'm not sure whats happening then. I run the situation and test 100 times, my seven devices (all running asterisk 1.4.43) 99.9% of the time join the conference. One in 100 times- one of my seven devices did not join the conference. I am trying to figure out why. Jerry -- _ -- 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
[asterisk-users] meetme race condition
I think I have a race condition. I am running something like this in my dialplan call agi to bring my list of devices into my MeetMe Playback beep start MeetMe() So in fact the meetme is not started before I bring the list of devices into the meetme. How can I do this differently so the MeetMe is started first or how can I wait in my AGI on the MeetMe to start because the MeetMe wont start until I exit the AGI... - or how do I in the dialplan wait for for the Meetme because I do have a stage where I redirect the Call into the MeetMe. so how do I inject a line that waits there for the MeetMe to be active??? THanks Jerry -- _ -- 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
[asterisk-users] MeetMe
I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20 in a Meetme. I can tell a difference (as two of the devices are close to each other) that they are not fully in sync. One was slightly behind the other... Any way to get them more in sync? Is it the delay from starting each device in the MeetMe? time to start device 1 till device X? I was expecting them to be all receiving the data for audio at the same time. Thanks, Jerry -- _ -- 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
- Original Message - From: Jerry Geis ge...@pagestation.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 1, 2012 5:01:43 PM Subject: [asterisk-users] MeetMe I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20 in a Meetme. I can tell a difference (as two of the devices are close to each other) that they are not fully in sync. One was slightly behind the other... Any way to get them more in sync? Is it the delay from starting each device in the MeetMe? time to start device 1 till device X? I was expecting them to be all receiving the data for audio at the same time. Nothing happens at the same time, unless you're broadcasting information over some transport that supports multicast sends. There's always going to be some interspersing of transmissions, if for no other reason than each participant's channel in the conference has to be serviced after the media has been mixed. With a sufficient number of participants, there will be some 'delay' between when the audio is sent to participant #1 to participant #n. Even if we assume that the transmissions are being done completely in parallel using multiple threads, you can still overcome the capabilities of a system by having a sufficiently large number of participants in a conference. That is, for any given system, you can always add more participants, such that, eventually, a thread will not be serviced immediately when it has data to send to a participant. A context switch will have to occur, resulting in the data being sent to said participant at a latter time, resulting in the work being done not at the same time. Note that recent versions of Asterisk (10+) have a revamped conferencing application (ConfBridge) that, in performance tests, performed much better than MeetMe. A big limitation of MeetMe is its reliance on DAHDI for mixing. ConfBridge removed this limitation, and typically can mix more participants at a faster rate. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
On Mon, 1 Oct 2012, Jerry Geis wrote: I am using Meeting on 1.4.43 with a handfull of devices, like 10 to 20 in a Meetme. I can tell a difference (as two of the devices are close to each other) that they are not fully in sync. You would have to measure how many ms they are 'out of sync' to determine if there is an issue or not. If you want a real eye-opener, try talking and listening to yourself with a cell phone on each ear. The delay can be several hundred ms and yet nobody ever complains because in the 'real world' we never listen to 2 endpoints at the same time. For future reference... A better subject may yield better answers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 not fully in sync
Nothing happens at the same time, unless you're broadcasting information over some transport that supports multicast sends. There's always going to be some interspersing of transmissions, if for no other reason than each participant's channel in the conference has to be serviced after the media has been mixed. With a sufficient number of participants, there will be some 'delay' between when the audio is sent to participant #1 to participant #n. Even if we assume that the transmissions are being done completely in parallel using multiple threads, you can still overcome the capabilities of a system by having a sufficiently large number of participants in a conference. That is, for any given system, you can always add more participants, such that, eventually, a thread will not be serviced immediately when it has data to send to a participant. A context switch will have to occur, resulting in the data being sent to said participant at a latter time, resulting in the work being done not at the same time. Note that recent versions of Asterisk (10+) have a revamped conferencing application (ConfBridge) that, in performance tests, performed much better than MeetMe. A big limitation of MeetMe is its reliance on DAHDI for mixing. ConfBridge removed this limitation, and typically can mix more participants at a faster rate. Mathew, Makes sense does it help that the system should not be mixing as I have set it up (I think) that it is PA only, there is no talk back. I am using options l and q in meetme. Should be listen only. Thanks Jerry -- _ -- 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 not fully in sync
- Original Message - From: Jerry Geis ge...@pagestation.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 1, 2012 8:01:06 PM Subject: Re: [asterisk-users] MeetMe not fully in sync Mathew, Makes sense does it help that the system should not be mixing as I have set it up (I think) that it is PA only, there is no talk back. I am using options l and q in meetme. Should be listen only. Thanks Nope, that doesn't make any difference. Disregarding the fact that the underlying structure of the application is unchanged, you still have n channels listening to some audio source. Each channel will have to have that audio sent on that channel. The fact that DAHDI doesn't have to do much in order to prepare the audio for those channels just means it has less 'mixing' related work to do. I may be missing something here, but - unless you have a business case that requires some functionality MeetMe provides - it sounds as if Page would be more appropriate than MeetMe. Its certainly a lighter weight approach to sending audio from a single source to multiple devices. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] meetme and dtmf
Hi Group, is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. Thanks for help Daniel -- _ -- 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 and dtmf
On Thu, 31 May 2012, Daniel Knoll wrote: is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. I'm just a 1.2 Luddite, but... You can use the meetme() 'X' option to jump out of the meetme and into another context. I use this to allow conference administrators to mute, un-mute, or kick users. The first digit jumps out of the meetme and into another context where I read additional digits (the user index) and then call an AGI (meetmeadmin-by-index) before returning the admin to the conference. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 identify user number
Daniel wrote: Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. If you have the user menu enabled, and the user is muted, then option 2 sets a 'Requests the Floor' flag. I know that the conference display feature in Web-MeetMe can interpret that flag and display a message that the caller would like to be unmated. I don't know of any other conference management apps that do, but I really have not looked into it. The request the floor feature was added in one of the early 1.6 releases, so unless you are on a truly ancient version, the backend support should be there. Dan -- _ -- 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
[asterisk-users] meetme identify user number
Hi Group, is in MeetMe any option to identify the own number (from the view of a caller)? I would like to write an option to set on the telephone an request for voice, if the room is muted. That request should display on our Conference Control Website and an Admin should unmute this person. Thanx for help. Daniel -- _ -- 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
[asterisk-users] meetme timeout if only one participant
Is it possible to have a meetme conference timeout (and go to the next line in the dialplan) if there is only one participant left? Thanks, Matt -- _ -- 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 timeout if only one participant
Don't think so. You can set up in the dialplan to skip meetme if the count is 0 or use meetmeadmin to kick out the user when he/she is the last one. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Tuesday, April 03, 2012 11:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme timeout if only one participant Is it possible to have a meetme conference timeout (and go to the next line in the dialplan) if there is only one participant left? Thanks, Matt -- _ -- 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
[asterisk-users] meetme with timerfd
We use MeetMe with res_timing_dahdi as the timing source, and once a while we get the following error which then causes Asterisk to crash/restart (with safe Asterisk). ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks According to the following from Asterisk wiki, DAHDI is required for MeetMe. Some confusion has arisen regarding the fact that non-DAHDI timing interfaces are available now. One common misconception which has arisen is that since timing can be provided elsewhere, DAHDI is no longer required for using the MeetMe application. Unfortunately, this is not the case. In addition to providing timing, DAHDI also provides a conferencing engine which the MeetMe application requires. I'm curious if DAHDI require/use res_timing_dahdi for it to run/function properly. Can we use res_timing_timerfd (instead of res_timing_dahdi) along with DAHDI for MeetMe? Thanks a lot, Matt -- _ -- 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
[asterisk-users] MeetMe or ConfBridge live meeting Streaming to the internet.
Hi All, Can someone please tell me if it is possible and if so how do I go about streaming a live conference to the internet for internet users to listen to? I'd hope to be able to do thus dynamically as conferences are created and internet users can tune in via a browser or streaming through media player. Regards David Klaverstyn -- _ -- 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 or ConfBridge live meeting Streaming to the internet.
On 07-03-12 11:44, David Klaverstyn wrote: Hi All, Can someone please tell me if it is possible and if so how do I go about streaming a live conference to the internet for internet users to listen to? I’d hope to be able to do thus dynamically as conferences are created and internet users can tune in via a browser or streaming through media player. Perhaps it's possible to stream the conference with Icecast? In 1.8.9.3 there is an ices module which allows you to stream audio from Asterisk to an Icecast server. Have you looked at that? Regards, Patrick -- _ -- 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 - Unable to write frame to channel
2012-01-20 20:09, Matt Hamilton skrev: Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? As you write I have seen this also with SIP in Meetme conferences sometimes when sip-channels is hung up. I havn't found any real problem or bad sound related to this, so I usually ignore this error. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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 - Unable to write frame to channel
I'm not using meetme directly - I'm using SLA which internally uses meetme and creates conferences for SLA trunks. There are no sound problems for me, either, but when the caller hangs up and this error occurs, the trunk statuses are not updated properly and the phones still show them as in use or hold. It's really hard to duplicate it - it seems to happen more under heavier load though. Matt Date: Sun, 22 Jan 2012 13:36:07 +0100 From: li...@jttech.se To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] meetme - Unable to write frame to channel 2012-01-20 20:09, Matt Hamilton skrev: Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? As you write I have seen this also with SIP in Meetme conferences sometimes when sip-channels is hung up. I havn't found any real problem or bad sound related to this, so I usually ignore this error. -- Johan Wilfer email: jo...@jttech.se JT Tech | Developer webb: http://jttech.se -- _ -- 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 -- _ -- 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
[asterisk-users] meetme - Unable to write frame to channel
Hi, Once in a while when a SIP channel connected to meetme conference is hung up, I start getting the following error multiple times: WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to channel Local/100203@h The status of the channel is not updated, and the only way to get back to normal is to restart Asterisk. Any thoughts? Is this a timing issue? Thanks a lot, Matt -- _ -- 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 with IVR
What version of Asterisk are you trying to implement this in? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta Sent: Tuesday, January 17, 2012 1:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme with IVR Any one is help ? Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.com wrote: Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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 with IVR
Asterisk 1.4.27 using . Best Regards, Mahesh Katta On Tue, Jan 17, 2012 at 8:01 PM, Danny Nicholas da...@debsinc.com wrote: What version of Asterisk are you trying to implement this in? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mahesh katta *Sent:* Tuesday, January 17, 2012 1:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] meetme with IVR ** ** Any one is help ? Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.com wrote: Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta ** ** -- _ -- 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 -- _ -- 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
[asterisk-users] meetme with IVR
Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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 with IVR
Any one is help ? Best Regards, Mahesh Katta On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.comwrote: Hi all, please help me. how we can configure between call add the IVR. My scenarios is A get the incomming call from C.In between them I need to one side IVR play for C, C enter the some DTMF inputs and A should be on hold. after finish C input will complete again they want talk each other .This is the scenario. Can anybody help to me how can I add this IVR in between those call, and how my asterisk will detect the DTMF input Best Regards, Mahesh Katta -- _ -- 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 does not return back to the dialplan
In article CA+xMSg43sQ=ichydct27dvbjgwkmot3npab0fc2m_libsrh...@mail.gmail.com,
Karim Mardhani ka...@vertexcommunication.ca wrote:
Hi everyone,
I am trying to get Meetme to return back to the context from where it
joined the meetme. For example a user uses the following context to join a
conference, once user hangs up I would like to continue executing the rest
of the dialplan. But when caller hangs up from the conference I see on CLI
that meetme exited with non-zero status but none of the rest of the
dialplan is executed. Please help. I am using asterisk 1.6.2.20
[default]
exten = _,1,MeetMe(1000,1pdMX)
exten = _,n,noop(returned from meetme) ;After user hangs up should
come here
exten = _,n,SoftHangup(${ORIG_CALLER})
exten = _,n,SoftHangup(${CONF_CALLER})
exten = _,n,Hangup
exten = h,1,noop(default-end)
exten = h,n,SoftHangup(${ORIG_CALLER})
exten = h,n,SoftHangup(${CONF_CALLER})
exten = h,n,Hangup
That's not how Asterisk works. When the caller hangs up, execution of
the current dialplan extension stops, and control passes to the 'h'
extension, if one exists in the current context.
Any processing you want to do when the caller hangs up must be done
in the 'h' extension.
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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[asterisk-users] Meetme does not return back to the dialplan
Hi everyone,
I am trying to get Meetme to return back to the context from where it
joined the meetme. For example a user uses the following context to join a
conference, once user hangs up I would like to continue executing the rest
of the dialplan. But when caller hangs up from the conference I see on CLI
that meetme exited with non-zero status but none of the rest of the
dialplan is executed. Please help. I am using asterisk 1.6.2.20
[default]
exten = _,1,MeetMe(1000,1pdMX)
exten = _,n,noop(returned from meetme) ;After user hangs up should
come here
exten = _,n,SoftHangup(${ORIG_CALLER})
exten = _,n,SoftHangup(${CONF_CALLER})
exten = _,n,Hangup
exten = h,1,noop(default-end)
exten = h,n,SoftHangup(${ORIG_CALLER})
exten = h,n,SoftHangup(${CONF_CALLER})
exten = h,n,Hangup
--
Karim Mardhani
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Re: [asterisk-users] meetme
hello, when i use the number of the first provider like that exten = 520870900,1,Answer exten = 520870900,n,Wait(4) exten = 520870900,n,Meetme All works without problem,the issue just with the second provider i use just the last 3 numbers for the outbound all works without issue, but whe i use the same 3 last numebrs for meetme i get the problem there is no result with outbond call ok exten = 527,1,Dial(SIP/223, 30) with meetme KO exten = 527,1,Answer exten = 527,n,Wait(4) exten = 527,n,Meetme please advice 2011/10/5 virendra bhati virbh...@gmail.com hi, you are using pattern matching and not using the right syntax like that. exten = _520,1,answer like that. On 5 Oct 2011 21:47, salaheddine elharit salah.elharit...@gmail.com wrote: Hello list i have one question related to meetme,i have to providers with the first one i put the number with 9 digit 520XX and all works without issue, with the second i put just the last 3 numbers 500 with meetme there is nothing but when i put the last 3 numbers like below i can call my sip without any problem, could you please inform me if the issue is related to my provider of the issue come from asterisk exten = 500,1,Dial(SIP/228, 30) extensions.conf first provider exten = 520XX,1,Answer exten = 520XX,n,Wait(4) exten = 520XX,n,Meetme = second provider exten = 500,1,Answer exten = 500,n,Wait(4) exten = 500,n,Meetme there is no meetme with this one meetme.conf conf =1234,5678 thanks and regards -- _ -- 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 -- _ -- 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
[asterisk-users] meetme
Hello list i have one question related to meetme,i have to providers with the first one i put the number with 9 digit 520XX and all works without issue, with the second i put just the last 3 numbers 500 with meetme there is nothing but when i put the last 3 numbers like below i can call my sip without any problem, could you please inform me if the issue is related to my provider of the issue come from asterisk exten = 500,1,Dial(SIP/228, 30) extensions.conf first provider exten = 520XX,1,Answer exten = 520XX,n,Wait(4) exten = 520XX,n,Meetme = second provider exten = 500,1,Answer exten = 500,n,Wait(4) exten = 500,n,Meetme there is no meetme with this one meetme.conf conf =1234,5678 thanks and regards -- _ -- 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
hi, you are using pattern matching and not using the right syntax like that. exten = _520,1,answer like that. On 5 Oct 2011 21:47, salaheddine elharit salah.elharit...@gmail.com wrote: Hello list i have one question related to meetme,i have to providers with the first one i put the number with 9 digit 520XX and all works without issue, with the second i put just the last 3 numbers 500 with meetme there is nothing but when i put the last 3 numbers like below i can call my sip without any problem, could you please inform me if the issue is related to my provider of the issue come from asterisk exten = 500,1,Dial(SIP/228, 30) extensions.conf first provider exten = 520XX,1,Answer exten = 520XX,n,Wait(4) exten = 520XX,n,Meetme = second provider exten = 500,1,Answer exten = 500,n,Wait(4) exten = 500,n,Meetme there is no meetme with this one meetme.conf conf =1234,5678 thanks and regards -- _ -- 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 not prompting for PIN
Doug Lytle wrote: I've been searching the Jira issue tracker and found a ticket: What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source and compiled it into my current Asterisk setup. I now have PIN prompts. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 not prompting for PIN
On Mon, Jul 11, 2011 at 8:22 AM, Doug Lytle supp...@drdos.info wrote: Doug Lytle wrote: I've been searching the Jira issue tracker and found a ticket: What I ended up doing was to copy the app_meetme.c out of the 1.4.30 source and compiled it into my current Asterisk setup. I now have PIN prompts. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. I was going to suggest either comparing the code or doing what you did. I hope you test it well, my guess is it should be fine. What is documented as far as the changelog between the two versions? That should give you an idea if there are security issues, or locking, or whatever. The code was changed for a reason. I thought the 1.4.X was EOL except for security (and possibly bug fixes?) -- _ -- 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 not prompting for PIN
That patch to 1.8 was a very simple change: modify one line, add another line. Should be easy and straight-forward to replicate on 1.4.42. (Not using 1.4 anymore over here, otherwise I would've provided the patch.) -- _ -- 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
[asterisk-users] Meetme not prompting for PIN
I've just put into place an updated meetme server. I went from 1.4.20.1
to 1.4.42.
In testing, it would seem that dynamically created conferences will not
prompt for the PIN. I've read though the readme and even went as far as
reading the 1.2 to 1.4 upgrade document.
s far as I can see, there has been no changes in the way dynamically
created conference rooms are handled. This is working correctly on a
1.4.30 server.
Below is a snippet of my dialplan with console output:
Dialplan:
exten =
s-process,n,Meetme(${conference.room}|ciMDPs|${conference.password})
Console:
-- Executing [s-process@mysql-meetme:1] SetMusicOnHold(DAHDI/1-1,
conference) in new stack
-- Executing [s-process@mysql-meetme:2] MeetMe(DAHDI/1-1,
1000|ciMDPs|1242) in new stack
-- Created MeetMe conference 1023 for conference '1000'
-- Recording
-- DAHDI/1-1 Playing 'vm-rec-name' (language 'en')
-- DAHDI/1-1 Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-1000-1 format: sln, 0xb742cd40
-- User ended message by pressing #
-- DAHDI/1-1 Playing 'auth-thankyou' (language 'en')
It skips prompting for password and moves right into prompting for the
recording of your name.
Suggestions would be appreciated,
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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Re: [asterisk-users] Meetme not prompting for PIN
On Sat, Jul 9, 2011 at 11:31 AM, Doug Lytle supp...@drdos.info wrote:
I've just put into place an updated meetme server. I went from 1.4.20.1 to
1.4.42.
In testing, it would seem that dynamically created conferences will not
prompt for the PIN. I've read though the readme and even went as far as
reading the 1.2 to 1.4 upgrade document.
s far as I can see, there has been no changes in the way dynamically
created conference rooms are handled. This is working correctly on a 1.4.30
server.
Below is a snippet of my dialplan with console output:
Dialplan:
exten = s-process,n,Meetme(${**conference.room}|ciMDPs|${**
conference.password})
Console:
-- Executing [s-process@mysql-meetme:1] SetMusicOnHold(DAHDI/1-1,
conference) in new stack
-- Executing [s-process@mysql-meetme:2] MeetMe(DAHDI/1-1,
1000|ciMDPs|1242) in new stack
-- Created MeetMe conference 1023 for conference '1000'
-- Recording
-- DAHDI/1-1 Playing 'vm-rec-name' (language 'en')
-- DAHDI/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/meetme/**meetme-username-1000-1
format: sln, 0xb742cd40
-- User ended message by pressing #
-- DAHDI/1-1 Playing 'auth-thankyou' (language 'en')
It skips prompting for password and moves right into prompting for the
recording of your name.
Suggestions would be appreciated,
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
I guess you could do it the old fashioned way until you open a ticket on
Bugtracker?
Sorry, not much help, but I believe that Bugtracker is the proper way to
report a bug.
Thanks,
Steve Totaro
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Re: [asterisk-users] Meetme not prompting for PIN
Steve Totaro wrote: I guess you could do it the old fashioned way until you open a ticket I've been searching the Jira issue tracker and found a ticket: https://issues.asterisk.org/jira/browse/ASTERISK-16747 Not being familiar with the new Jira system, I can't seem to find a patch for the 1.4 series, if I'm reading it correctly, the fix only went into 1.8. Can someone review and confirm? Thanks! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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 Time Limit?
Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- _ -- 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 -- _ -- 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 Time Limit?
On Thu, Apr 21, 2011 at 4:03 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- Dhaval's reply works for when you're running a MeetMe conference directly, which does not help Bryant (the question was phrased a little oddly, which caused the confusion I think) Regarding how to limit how long the Paging call can be, use the TIMEOUT(absolute) function. Here's an AEL example: [paging] exten = _92XX,1,Noop(Making sure the call only lasts 60 seconds or less) same = n,Set(TIMEOUT(absolute)=60); same = n,Page(insert page targets and options) Let me know if that works out for you! Regarding MeetMe time limiting in general, I'd like to add an alternative to Dhaval's solution, just to get it back out there in the intertubes so people can find it in the future. As of Asterisk 1.6 you can schedule RealTime MeetMe conferences. I've attached a structure dump of a table called conferences, just direct your extconfig.conf to use it for meetme, set schedule=yes in meetme.conf, and then set the start and end times in the table when creating a scheduled conference. Cheers all! Sherwood McGowan Coming soonSamuPBX scheduled_conferences.sql Description: Binary data -- _ -- 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
[asterisk-users] Meetme Time Limit?
Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- _ -- 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 headache
hey just change following [status-one-en] exten = 100,1,Meetme (12345,qdM) exten = 100,1,Hangup() Channel: Local/100@status-one-en CallerID: Rick 55 MaxRetries: 0 RetryTime: 15 WaitTime: 45 Application: Playback Data: my_status_message On Mon, Apr 4, 2011 at 10:38 PM, D. Rick Anderson rander...@customteleconnect.com wrote: Ok, I've been running applications on 1.4 for quite some time using meetme to hold a person, while the person on the other end of the call accepts, etc. I was playing status messages to the calling party using a context like this: [status-one-en] exten = 100,1,Playback(my_status_message) exten = 100,1,Hangup() and then creating a call file like this: Channel: Local/100@status-one-en CallerID: Rick 55 MaxRetries: 0 RetryTime: 15 WaitTime: 45 Application: MeetMe Data: 12345,qdM and it would hook into the meetme, play the message, then hangup and drop out. I've been building an application with 1.6, and this isn't working at all. In verbose mode, I see the message played, and the call hang up, but the music never even stops on the meetme. After about 20 seconds I get: Call failed to go through, reason (3) Remote end Ringing Is there some other way to do this in 1.6 that I'm unaware of? I've tried creating a context and extension for the meetme portion (rather than using the Application/Data in the call file, and switched the order around (which does cause the music to stop, but the announcement still doesn't get played, and I get the same call failed message). I've been googling on this for days now, and really just need to get it working. TIA Rick CONFIDENTIALITY / PRIVILEGE NOTICE: This transmission and any attachments are intended solely for the addressee. This transmission is covered by the Electronic Communications Privacy Act, 18 U.S.C §§ 2510-2521. The information contained in this transmission is confidential in nature and protected from further use or disclosure under U.S. Pub. L. 106-102, 113 U.S. Stat. 1338 (1999), and may be subject to attorney-client or other legal privilege. Your use or disclosure of this information for any purpose other than that intended by its transmittal is strictly prohibited, and may subject you to fines and/or penalties under federal and state law. If you are not the intended recipient of this transmission, please DESTROY ALL COPIES RECEIVED and confirm destruction to the sender via return transmittal. -- _ -- 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 -- _ -- 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
[asterisk-users] MeetMe headache
Ok, I've been running applications on 1.4 for quite some time using meetme to hold a person, while the person on the other end of the call accepts, etc. I was playing status messages to the calling party using a context like this: [status-one-en] exten = 100,1,Playback(my_status_message) exten = 100,1,Hangup() and then creating a call file like this: Channel: Local/100@status-one-en CallerID: Rick 55 MaxRetries: 0 RetryTime: 15 WaitTime: 45 Application: MeetMe Data: 12345,qdM and it would hook into the meetme, play the message, then hangup and drop out. I've been building an application with 1.6, and this isn't working at all. In verbose mode, I see the message played, and the call hang up, but the music never even stops on the meetme. After about 20 seconds I get: Call failed to go through, reason (3) Remote end Ringing Is there some other way to do this in 1.6 that I'm unaware of? I've tried creating a context and extension for the meetme portion (rather than using the Application/Data in the call file, and switched the order around (which does cause the music to stop, but the announcement still doesn't get played, and I get the same call failed message). I've been googling on this for days now, and really just need to get it working. TIA Rick CONFIDENTIALITY / PRIVILEGE NOTICE: This transmission and any attachments are intended solely for the addressee. This transmission is covered by the Electronic Communications Privacy Act, 18 U.S.C §§ 2510-2521. The information contained in this transmission is confidential in nature and protected from further use or disclosure under U.S. Pub. L. 106-102, 113 U.S. Stat. 1338 (1999), and may be subject to attorney-client or other legal privilege. Your use or disclosure of this information for any purpose other than that intended by its transmittal is strictly prohibited, and may subject you to fines and/or penalties under federal and state law. If you are not the intended recipient of this transmission, please DESTROY ALL COPIES RECEIVED and confirm destruction to the sender via return transmittal. -- _ -- 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 conference playback of random sound file
Check out the Random Application and the RAND function, Here is a
quick untested example for either.
exten = s,1,Answer
exten = s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten = s,n,Random(33:${CONTEXT},s,FILE1) ; 33% Num1
exten = s,n,Random(33:${CONTEXT},s,FILE2) ; 33% Num2
exten = s,n,Random(34:${CONTEXT},s,FILE3) ; 34% Num3
exten = s,n(FILE1),Background(tt-monkeys)
exten = s,n,Goto(Connect)
exten = s,n(FILE2),Background(tt-weasels)
exten = s,n,Goto(Connect)
exten = s,n(FILE3),Background(gambling-drunk)
exten = s,n,Goto(Connect)
exten = s,n(CONNECT),NoOp
exten = s,n,Meetme(options)
Or using RAND if your prompts are all numbered as prompt0 to prompt100
exten = s,1,Answer
exten = s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten = s,n,Set(promptnum=${RAND(1,100)})
exten = s,n,Background(prompt${promptnum})
exten = s,n,Meetme(options)
On Thu, Feb 10, 2011 at 5:58 PM, John Jolly jgjo...@gmail.com wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a
random sound file from a specific directory prior to it dropping the caller
into the conference itself. i am able to successfully get it to play a
specific file prior to entering the conference unsure how to implement this
sort of randomization.
Is this possible? Any help will be greatly appreciated.
john jolly jgjolly[at]gmail.com
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[asterisk-users] meetme conference playback of random sound file
i have been trying to find a way to accomplish the following but have not found a method in which to do so: i am trying to configure the meetme conference (asterisk 1.8) to play a * random* sound file from a specific directory prior to it dropping the caller into the conference itself. i am able to successfully get it to play a specific file prior to entering the conference unsure how to implement this sort of randomization. Is this possible? Any help will be greatly appreciated. john jolly jgjolly[at]gmail.com -- _ -- 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 conference playback of random sound file
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote: i am trying to configure the meetme conference (asterisk 1.8) to play a * random* sound file from a specific directory prior to it dropping the caller into the conference itself. Absent an Asterisk-specific solution, how about a separate process which would link a random file into a fixed pathname? (Fired off from cron, perhaps.) Roger -- _ -- 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 conference playback of random sound file
On Thu, 10 Feb 2011, John Jolly wrote: i am trying to configure the meetme conference (asterisk 1.8) to play a random sound file from a specific directory prior to it dropping the caller into the conference itself. i am able to successfully get it to play a specific file prior to entering the conference unsure how to implement this sort of randomization. Who is the sound file played to? The caller or the conference? Please show what you are using now. Would an AGI that selected a random file from the directory and set the path as a channel variable work? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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 and admin users
In article 1296748085.2237.16.camel@shaft, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Is there an option on MeetMe that means the conference room is only available if an admin user is logged in? I've had a look the the application from the asterisk cli but I can't really see what I'm after. Currently using 1.4.17 (deb package) Soon moving up to 1.8.2 (rpm package) What you do is give admin users the A flag (marked user) as well as the a flag (admin user). Then you also give all users the w flag (wait until marked user joins) and optionally the x flag (exit when all marked users have left). Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- 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 and admin users
On Thu, 2011-02-03 at 16:39 +, Tony Mountifield wrote: In article 1296748085.2237.16.camel@shaft, Ishfaq Malik i...@pack-net.co.uk wrote: Hi Is there an option on MeetMe that means the conference room is only available if an admin user is logged in? I've had a look the the application from the asterisk cli but I can't really see what I'm after. Currently using 1.4.17 (deb package) Soon moving up to 1.8.2 (rpm package) What you do is give admin users the A flag (marked user) as well as the a flag (admin user). Then you also give all users the w flag (wait until marked user joins) and optionally the x flag (exit when all marked users have left). Cheers Tony Thanks Tony That makes sense, however, I have a problem that this is on an incoming real phone number, but I'm sure I can work something out now I know the underlying principle. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 - ConfBridge: hint not working
On 12/21/2010 10:15 PM, sean darcy wrote:
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
core show version
Asterisk 1.6.2.16-rc1
sean
BTW, wasn't device state handling added to ConfBridge last March?
https://issues.asterisk.org/view.php?id=16972
sean
Entered as:
https://issues.asterisk.org/view.php?id=18518
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[asterisk-users] MeetMe - ConfBridge: hint not working
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
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Re: [asterisk-users] MeetMe - ConfBridge: hint not working
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
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Re: [asterisk-users] MeetMe - ConfBridge: hint not working
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
core show version
Asterisk 1.6.2.16-rc1
sean
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Re: [asterisk-users] MeetMe - ConfBridge: hint not working
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten = 81,hint,ConfBridge:81
;;exten = 82,hint,MeetMe:82
exten = 82,hint,ConfBridge:82
;;exten = 83,hint,MeetMe:83
exten = 83,hint,ConfBridge:83
;;exten = 84,hint,MeetMe:84
exten = 84,hint,ConfBridge:84
And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge?
sean
core show version
Asterisk 1.6.2.16-rc1
sean
BTW, wasn't device state handling added to ConfBridge last March?
https://issues.asterisk.org/view.php?id=16972
sean
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Re: [asterisk-users] Meetme and MOH
Thanks all, I realised after posting 2 things.. 1) I needed to also cover MOH outside of meetme. And that 2) theres a bug in 1.4.18 where the defaults aren't reloaded properly for MOH, and you have to do a server stop/start to get them to reload. Thanks, Adrian -- _ -- 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 and MOH
Adrian Marsh wrote: Thanks all, I realised after posting 2 things.. 1) I needed to also cover MOH outside of meetme. And that 2) theres a bug in 1.4.18 where the defaults aren't reloaded properly for MOH, and you have to do a server stop/start to get them to reload. Thanks, Adrian Probably why there is a 1.4.37? I found many things broken between 1.4.13 and 1.4.21 But that is now ancient history John Novack -- Dog is my Co-pilot -- _ -- 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 and MOH
Yes John... but I also now find in testing many things broken between my IAX provider and 1.4.37 Which is a reason to hold back... Thanks, Adrian From: John Novack [mailto:jnov...@stromberg-carlson.org] Sent: 26 November 2010 13:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Adrian Marsh Subject: Re: [asterisk-users] Meetme and MOH Adrian Marsh wrote: Thanks all, I realised after posting 2 things.. 1) I needed to also cover MOH outside of meetme. And that 2) theres a bug in 1.4.18 where the defaults aren't reloaded properly for MOH, and you have to do a server stop/start to get them to reload. Thanks, Adrian Probably why there is a 1.4.37? I found many things broken between 1.4.13 and 1.4.21 But that is now ancient history John Novack -- Dog is my Co-pilot -- _ -- 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
[asterisk-users] Meetme Realtime in 1.6
I have a server running 1.6.2.13 that uses realtime for most configurations. Everything works fine except for meetme. When I use Meetme with Realtime any options specified in the dial plan are ignored. For example: exten = 1557,1,Meetme(905,icM(somemusic)) With realtime I just get dropped into the conference room. If I used meetme.conf directly it does prompt me for my name and use the proper MoH. Should I open a bug for this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 Realtime in 1.6
Hi Carlos, you have to incllude the conference options (user ad admin) in the meetme table and put schedule=yes in meetme.conf file On the dialplan just call the conference like: exten = 1557,1,Meetme(905) Regards - Bakko -- _ -- 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
[asterisk-users] Meetme and MOH
Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than default ? Asterisk: 1.4.15 Thanks, Adrian -- _ -- 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 and MOH
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh adrian.ma...@ubiquisys.com wrote: Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than “default” ? Asterisk: 1.4.15 Thanks, Adrian -- _ -- 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 Use Set(CHANNEL(musicclass)=MUSICONHOLDCLASSYOUWANT). What I do is add a column to the conferences/meetme table in my database to hold the moh class I want and then retrieve that in the dialplan use the aforementioned command. -- _ -- 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 and MOH
On Thu, Nov 18, 2010 at 12:37 PM, Adrian Marsh adrian.ma...@ubiquisys.comwrote: Hi, With a dynamic Meetme using: MeetMe(|DsMrc) How do I control which context MOH uses, other than “default” ? Asterisk: 1.4.15 In 1.4.x you would use SetMusicOnHold(class) before you called your MeetMe() in the dialplan. In 1.6.x (at least 1.6.2.x), you would use Set(CHANNEL(musicclass)=...) instead. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- 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
[asterisk-users] Meetme
Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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
Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko-- _ -- 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
hi Bakko, thanks! Acctualy, I had tried this but still don´t works! [conference]exten = 1001,3,MeetMe(1001,ipdM)exten = 1001,4,Set(CHANNEL(language)=pt_BR)exten = 1001,5,Playback(pt_BR/vm-goodbye)exten = 1001,6,Hangup this is my config! What´s wrong? thanks again! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: asannu...@gmail.com To: asterisk-users@lists.digium.com Date: Sun, 17 Oct 2010 16:36:34 -0500 Subject: Re: [asterisk-users] Meetme Hi Flavio, try with this funtion before the line with the english meetme application Set(CHANNEL(language)=en) and Set(CHANNEL(language)=pr) before the line with the portugues meetme application Regards - Bakko -- _ -- 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 -- _ -- 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
