Re: [asterisk-users] conferencing without DAHDI
Am 08.02.2010 21:15, schrieb Philippe Sultan: Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? Yes. Announcements are played to the conference members by creating a channel of type 'Bridge' which streams the sound files. thanks klaus Further, is there somewhere a documentation Well, there is no sample configuration in the tarball because ConfBridge does require any configuration file. I wonder what mute should mean. Does it mean that the participant will not receive any media, or that media sent by the participant will be ignored, or both? thanks klaus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: I wonder what mute should mean. Does it mean that the participant will not receive any media, or that media sent by the participant will be ignored, or both? Please post your discoveries to: http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge so we can all learn at together. I wrote that up when I couldn't find documentation on my own. Obviously it's short on details. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
Am 09.02.2010 15:35, schrieb David Backeberg: On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: I wonder what mute should mean. Does it mean that the participant will not receive any media, or that media sent by the participant will be ignored, or both? Answering myself: muting means that the participants voice is ignored. Please post your discoveries to: http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge done klaus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
On Tue, Feb 9, 2010 at 10:26 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Answering myself: muting means that the participants voice is ignored. Thank you for updating the wiki and the list. I looked into this when I was having problems with early 1.6.0.* MeetMe(), specifically the talker detection problem where beginnings and endings of sentences would be clipped and not mixed (exacerbated by SIP vad). I was able to tune MeetMe() and SIP better and solve my problem, but ConfBridge() certainly seemed promising. A big thanks to Josh Colp and others for making this happen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conferencing without DAHDI
Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
And by the way, app_confbridge is included in the 1.6.2 series (at least). On Mon, Feb 8, 2010 at 1:49 PM, Philippe Sultan philippe.sul...@gmail.com wrote: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johansson o...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Philippe Sultan -- Philippe Sultan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
Hi Philippe! Am 08.02.2010 13:49, schrieb Philippe Sultan: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. That sounds good. thanks klaus The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johanssono...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
Am 08.02.2010 13:49, schrieb Philippe Sultan: Hi Klaus, The module is app_confbridge, and the application is ConfBridge. I had been using it for a while because it's really easy to use : you don't need any configuration file, and you get cool announcements upon conference events from a playback channel. Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? thanks klaus Further, is there somewhere a documentation The options work pretty much like meetme, although I would have liked to have a 'x' option to close the conference when the last marked user leaves. Moreover, I couldn't have the playback channel speak French, from what I've read in the source code, I think that feature would require a configuration file because the playback channel is not a per user option. Philippe On Mon, Feb 8, 2010 at 12:56 PM, Olle E. Johanssono...@edvina.net wrote: 8 feb 2010 kl. 12.29 skrev Klaus Darilion: Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. In Asterisk trunk there's a new conference bridge module you can test. There are also some third-party modules out there, like app_conference. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing without DAHDI
Philippe, what exactly is a playback channel? Is it a pseudo participant playing back the announcements? Yes. Announcements are played to the conference members by creating a channel of type 'Bridge' which streams the sound files. thanks klaus Further, is there somewhere a documentation Well, there is no sample configuration in the tarball because ConfBridge does require any configuration file. 'core show application ConfBridge' in the CLI will give you the options list. You'd probably also want to take a look at the app_confbridge.c file. Very short and readable for such a powerful app. Philippe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing and web front-end
would somebody be able to recommend a good package that works with Asterisk please ... not commercial as will be mainly used for my home office. Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing Hardware
On Mon, 29 Sep 2008, Jim Boykin wrote: Thanks Gordon Mike for the response. What accuracy are you getting from zaptest/dahdi_test (and system info). Two more questions: 1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel. 2) What about CPU load? I run a custom compiled kernel with the high resolution timers HPET. CPU load for me is next to nothing. zttest for me: # zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% ^C --- Results after 17 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.996410 System was just carrying a few SIP - IAX calls at that point. Gordon Thanks Jim On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL PROTECTED] wrote: Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing Hardware
We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. Thanks Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing Hardware
On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing Hardware
Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing Hardware
Thanks Gordon Mike for the response. What accuracy are you getting from zaptest/dahdi_test (and system info). Two more questions: 1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel. 2) What about CPU load? Thanks Jim On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL PROTECTED] wrote: Go for it. ztdummy is not an issue. I have used ztdummy with 220 simultaneous participants in 18 different conference groups. At one time, I had 60 machines running simultaneously in a FARM all of which were carrying the same 18 conference groups with over 200 participants active on each machine. ..mike.. At 11:23 AM 9/28/2008, Gordon Henderson wrote: On Sun, 28 Sep 2008, Jim Boykin wrote: We plan to use asterisk for conferencing. As I understand, it requires either a separate hardware like x100p clone or ztdummy. What are the pro cons of x100p vs ztdummy. Any other hardware suggestions for conferencing? It should be able to handle few simultaneous conferences. I have one server which handles a few simultaneous conferences using just ztdummy - however there are rarely more than 4-5 participants and rarely more than 2 conferences on the go at any one time.. (2.5GHz AMD Semperon FWIW) Ztdummy using: ztdummy: Trying to load High Resolution Timer ztdummy: Initialized High Resolution Timer ztdummy: Starting High Resolution Timer ztdummy: High Resolution Timer started, good to go And zttest gets more 100%'s than not. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing..
Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? What do you mean when you describe cards as having or not having Zaptel? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing..
Hi Ajey, which kind of BRI are you using? Giorgio Incantalupo Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing..
You can do conferencing without the zap interface. just modprobe ztdummy. Its good for small conferences. On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing..
On Mon, 14 Apr 2008, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. It can do conferencing without a zap interface too. The BRI cards I have they do not have Zaptel. And? How do I enable conferencing on my server? Well you could start by reading the manual, or books on the subject. There's a very good one avalable free for download too. Hint: You're looking for the MeetMe application. You need to read the book; Asterisk the future of telephony. Google for it, you can get it as a PDF. Which will save (as someone else here mentioned recently!) using up newbie karma points... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conferencing..
Hi Faraz, yes, you can use ztdummy but it cannot completely replace Digium cards. It depends from your hardwareI had troubles with some kind of serversso beware. Giorgio. Faraz R. Khan wrote: You can do conferencing without the zap interface. just modprobe ztdummy. Its good for small conferences. On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing..
I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Yeah that's what I thought. Am just trying to remember what caused it though. Maybe Tzafrir will chime in :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhIkgDQNt8rg0Kp4RAsY2AKCft6fPiWHgBtdE7dS3FpeGRQqnxACdF/Ee tSjBUM1DzdI1XzSVjRxlJ4s= =SMN2 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Well, yes. Is ztdummy loaded? cat /proc/zaptel/* What kernel version do you use? What version of Zaptel? What Linux distribution? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Tzafrir, cat /proc/zaptel/* Span 1: ZTDUMMY/1 ZTDUMMY/1 1 Kernel: 2.6.18-5-686 #1 SMP Zaptel: zaptel-1.2.20.1 OS: Debian GNU/Linux 4.0 i downgraded my zaptel from 1.2.22.1 to 1.2.20.1 but still the same. thanks again regards, nhadie Tzafrir Cohen wrote: On Wed, Jan 09, 2008 at 03:36:06PM +0800, Nhadie wrote: Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Well, yes. Is ztdummy loaded? cat /proc/zaptel/* What kernel version do you use? What version of Zaptel? What Linux distribution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en *CLI load chan_zap.so Unable to load module chan_zap.so -- on the log file it says, it as already loaded that's why it's unable to load. i tried my calling to my conf 6000 -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- Recording -- Playing 'vm-rec-name' (language 'en') -- Executing Set(SIP/100-081825b0, MEETME_OPTS=iM) in new stack -- Executing Goto(SIP/100-081825b0, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/100-081825b0, 6000|iM|) in new stack == Parsing '/etc/asterisk/meetme.conf': Found == Parsing '/etc/asterisk/meetme_additional.conf': Found -- Created MeetMe conference 1023 for conference '6000' -- Recording -- Playing 'vm-rec-name' (language 'en') it's trying to play something 'vm-rec-name' but i cannot hear anything on the phone. i'm using g711. i'm not using trixbox, i just installed asterisk, freepbx, zaptel, etc on a debian box. i'm using all the latest version i downloaded from the website (i used asterisk 1.2). /usr/include# modprobe -l | grep ztdum /lib/modules/2.6.18-5-686/misc/ztdummy.ko /usr/include# modprobe -l | grep zap /lib/modules/2.6.18-5-686/misc/zaptel.ko how do i know if my ztdummy is working properly? thanks again! regards, nhadie Steve Edwards wrote: dave cantera wrote: nhadie, meetme requires a zaptel timing device... ztdummy is unreliable when using meetme conferencing. On Wed, 9 Jan 2008, Nhadie wrote: hi dave thank you for the reply. i have loaded zap and using only ztdummy but still can't hear anything when i dial ti my conference, i think this explains it already. will a sangoma card do? I use ztdummy with meetme conferencing and it works fine on CentOS 4.5. Ztdummy is not an issue until you get xx callers in xx conferences. I think (but have no empirical data to back it up) that a card yields better sound quality at higher call levels. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en That means the zap channel should be ok. One thing you could do is go to the place you downloaded Zaptel and type: ./zttest -v Do you get numbers (i.e. something close or closish to 100%)? Also, if you just have the extensions: exten = 555,1,Answer() exten = 555,n,Background(demo-echotest) exten = 555,n,Echo() Do you get an answer? You don't really need the brackets on answer and echo but I usually type that way and then add options. :-) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhFUKDQNt8rg0Kp4RAtfQAKC/mjeswAVxnkzv/HHC/4ZCL92SEwCfRoY7 rIAGfpE/0dh56i9myEbOFfA= =fHxG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
I will be out of the office on Wednesday, January 9, 2008. If this is an emergency, please call Customer Service at (877) 791-7700. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Matt, I tried /usr/local/src/zaptel-1.2.22.1# ./zttest -v and it just freezes at this. Opened pseudo zap interface, measuring accuracy... no more outputs, when i cancelled this is what i got. --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 does that mean my zaptel is bad? Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Steve, I see. I have this now, *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault en That means the zap channel should be ok. One thing you could do is go to the place you downloaded Zaptel and type: ./zttest -v Do you get numbers (i.e. something close or closish to 100%)? Also, if you just have the extensions: exten = 555,1,Answer() exten = 555,n,Background(demo-echotest) exten = 555,n,Echo() Do you get an answer? You don't really need the brackets on answer and echo but I usually type that way and then add options. :-) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhFUKDQNt8rg0Kp4RAtfQAKC/mjeswAVxnkzv/HHC/4ZCL92SEwCfRoY7 rIAGfpE/0dh56i9myEbOFfA= =fHxG -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Hi Matt, it seems i don't have that command. *CLI zap show channels No such command 'zap' (type 'help' for help) *CLI ! abort add ael agent agi cdr databasedebug dnsmgr dontdump dundi extensions feature group helpiax2include indication initloadlocal logger meetme mgcp mixmonitor moh no realtimereload remove restart rtp set showsip skinny soft stopunload *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no -- Executing NoOp(SIP/104-58ae, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-58ae, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-58ae, 0?USER) in new stack -- Executing Answer(SIP/104-58ae, ) in new stack -- Executing Wait(SIP/104-58ae, 1) in new stack -- Executing Set(SIP/104-58ae, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-58ae, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-58ae, 6000||) in new stack Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. Can you do a zap show channels in the Asterisk console (without the ) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh JEjcAt3QDqV3aN0rAZGNq9g= =Zqs+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, it seems i don't have that command. :) You'll need to make sure that: 1. You have zaptel compiled 2. You compile Asterisk *after* zaptel is compiled and installed 3. You have either modprobed zaptel + ztdummy or made the service and started it. You didn't say, is this a straight Asterisk machine or trixbox/freepbx? If those are done and it still doesn't work then you can report the errors you get when you type (in the console): module load chan_zap.so - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgw3pDQNt8rg0Kp4RAvJGAKCtI+GaFMCcNk/PB1VMoyOo67RAwACeM5pJ BH0EhGK4hD+oL7TXu0d33+M= =1HK0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
Then it's time to build zaptel, then rebuild asterisk later, PaulH On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote: Hi Matt, it seems i don't have that command. *CLI zap show channels No such command 'zap' (type 'help' for help) *CLI ! abort add ael agent agi cdr databasedebug dnsmgr dontdump dundi extensions feature group helpiax2include indication initloadlocal logger meetme mgcp mixmonitor moh no realtimereload remove restart rtp set showsip skinny soft stopunload *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no *CLI show channeltypes TypeDescriptionDevicestate Indications Transfer -- ------ --- Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Local Local Proxy Channel Driver no yes no Skinny Skinny Client Control Protocol no yes no Phone Standard Linux Telephony API D no no no SIP Session Initiation Protocol (S yes yes yes IAX2Inter Asterisk eXchange Driver yes yes yes MGCPMedia Gateway Control Protocol no yes no -- Executing NoOp(SIP/104-58ae, Using CallerID 104 104) in new stack -- Executing Set(SIP/104-58ae, MEETME_ROOMNUM=6000) in new stack -- Executing GotoIf(SIP/104-58ae, 0?USER) in new stack -- Executing Answer(SIP/104-58ae, ) in new stack -- Executing Wait(SIP/104-58ae, 1) in new stack -- Executing Set(SIP/104-58ae, MEETME_OPTS=) in new stack -- Executing Goto(SIP/104-58ae, STARTMEETME|1) in new stack -- Goto (from-internal,STARTMEETME,1) -- Executing MeetMe(SIP/104-58ae, 6000||) in new stack Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: hi shane, thanks for your reply. i actually tried 3 phones dialled to the conference, but cant here anything from those phones. i also enabled the usercount so i can hear something at least. but still no sound. i'm using ztdummy, as i dont have a card yet. Can you do a zap show channels in the Asterisk console (without the ) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh JEjcAt3QDqV3aN0rAZGNq9g= =Zqs+ -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
On Tue, Jan 08, 2008 at 06:45:14PM +1300, Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nhadie wrote: Hi Matt, it seems i don't have that command. :) You'll need to make sure that: 1. You have zaptel compiled 2. You compile Asterisk *after* zaptel is compiled and installed 3. You have either modprobed zaptel + ztdummy or made the service and started it. In other words, what is the output of the following command from the Asteris CLI: module load chan_zap.so (This tests both cases right away: gives different error messages in the different cases) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferencing help
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: (This tests both cases right away: gives different error messages in the different cases) Sweet :) - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHgyysDQNt8rg0Kp4RAn6TAJ95CZiwFSgt8Vp+KKm/SOzfkJzi7QCgvSbO KgdOHiu1dEbD4qJ2BfTfqsY= =1zsg -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing Phones ...
Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conferencing Phones ...
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, February 09, 2007 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Conferencing Phones ... Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) I have an IP 4000, and I think the quality is excellent (on par with the analogs, which I also consider quite good). Most of our deployments continue to use fxs ports on a channel bank and analog phone, but that's mostly because we have a large investment in them. - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conferencing Phones ...
We use the Polycom soundstation 2W plugged into an iaxy...works very well... Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Friday, February 09, 2007 8:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Conferencing Phones ... Anyone got any experiences of good quality VoIP conferencing phones? I've used Polycom analogue units in the past, and I see that they have a SIP version (the IP4000) - but it is better/worse/as good as an analogue version? (ie. would I be better off with an analogue version into a TDM card or ATA?) Cheers, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conferencing Issue please help
Hi All, I have a problem in configuring in asterisk. I configure asterisk meetme.conf and extension.conf, but when i transfer call to conference it give me this message and asterisk kill it self. Ouch ... error while writing audio data: : Broken pipe If any one knows about this please help me to fix this. I'm using Asterisk 1.2.13 Thank You, Ishanka Anuradha Ranasooriya ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing with multiple servers
In general, you are talking of distributed conferencing, which in SIP it was tried once to standardize but never reached anything. It is just not commercially popular, i guess. Now, this doesn't mean that it cannot be done or that it has not been done ... but it is propietary implementations. And in my particular knowledge, i know asterisk was not the choice. Just my 2 cents. Cesc On 6/20/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 12:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Conferencing with multiple servers On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote: Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. This is a general problem with the 'enterprise grade' aspects of Asterisk. As far as I know, there is no way to distribute applications (eg: Queue, Meetme etc) between multiple Asterisk systems. You really need to run the applications that will serve a common set of phones on the same Asterisk system, and then fail over to a secondary if necessary. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing with multiple servers
Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. Any ideas on how to impliment this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing with multiple servers
On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote: Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. Any ideas on how to impliment this? Nope, never tried but I guess you could use something like app_bridge to do it. Check out http://bugs.digium.com/view.php?id=5841 Maybe you can also do it with the manager interface (search for AMI on voip-info.org). Good luck and let us know how it goes. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conferencing with multiple servers
-Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 12:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Conferencing with multiple servers On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote: Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. This is a general problem with the 'enterprise grade' aspects of Asterisk. As far as I know, there is no way to distribute applications (eg: Queue, Meetme etc) between multiple Asterisk systems. You really need to run the applications that will serve a common set of phones on the same Asterisk system, and then fail over to a secondary if necessary. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing without Meetme
I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not using the native bridging capabilities of the channels because apparently they only allow two channels. Is there any special precaution that I have to be aware of when doing this? Do I have to masquerade the channels that are inserted into the conference? The channels will mainly use SIP (maybe IAX2 too occasionally). Thanks for your help. -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing without Meetme
On Mon, 7 Feb 2005, Juan Jose Comellas wrote: I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. Isn't this similar to how app_conference works? See http://cvs.sourceforge.net/viewcvs.py/iaxclient/app_conference/ . Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing needs Zaptel ??
Hi All, This are going nice with my setup, but I m now trying to play with conferencig. My setup uses a isdn4linux card with no zaptel drivers. When trying configs for conferencing I m getting this errors bellow . What do I need to do to install this pseudo drivers ? Thanks in advance, Paulo Nov 12 13:33:52 WARNING[1110514608]: chan_zap.c:755 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 12 13:33:52 ERROR[1110514608]: chan_zap.c:6663 chandup: Unable to dup channel: No such device or address Nov 12 13:33:52 WARNING[1110514608]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 12 13:33:52 WARNING[1110514608]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') -- Playing 'conf-getconfno' (language 'en') Nov 12 13:34:06 WARNING[1110514608]: chan_zap.c:755 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 12 13:34:06 ERROR[1110514608]: chan_zap.c:6663 chandup: Unable to dup channel: No such device or address Francisco Paulo Adriano WaveLIS LDA Mobile +351 91 870 87 98 Office + 351 21 989 83 34 Fax +351 21 989 83 35 E-mail : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing needs Zaptel ??
Dear Paulo , You need *ztdummy* for conferencing (Meetme). -Jefferson Carvalho Paulo Adriano wrote: Hi All, This are going nice with my setup, but I m now trying to play with conferencig. My setup uses a isdn4linux card with no zaptel drivers. When trying configs for conferencing I m getting this errors bellow . What do I need to do to install this pseudo drivers ? Thanks in advance, Paulo Nov 12 13:33:52 WARNING[1110514608]: chan_zap.c:755 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 12 13:33:52 ERROR[1110514608]: chan_zap.c:6663 chandup: Unable to dup channel: No such device or address Nov 12 13:33:52 WARNING[1110514608]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 12 13:33:52 WARNING[1110514608]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') -- Playing 'conf-getconfno' (language 'en') Nov 12 13:34:06 WARNING[1110514608]: chan_zap.c:755 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 12 13:34:06 ERROR[1110514608]: chan_zap.c:6663 chandup: Unable to dup channel: No such device or address Francisco Paulo Adriano WaveLIS LDA Mobile +351 91 870 87 98 Office + 351 21 989 83 34 Fax +351 21 989 83 35 E-mail : [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing -- app_meetme, app_meetme2, app_conference
I want to build a conferencing system and I'm looking for suggestions on which application will make a better starting point. A conference consists of: up to 20 callers, a small number (1-3) of agents and a small number of supervisors (0-3). Multiple conferences will be active on the same host An agent's job is to keep the conference going. A supervisor's job is to keep the agents going :) Some of the characteristics I've identified are: 1) A supervisor may join and leave conferences at will without any indication to agents or callers. 2) A supervisor should be able to un-mute and mute themselves. 3) A supervisor should be able to coach an agent. In this mode, the supervisor can speak, but only the agent(s) can hear the supervisor. This will not interfere with the interaction between agent(s) and callers. 4) A supervisor should be able to kick an agent. At this point, the supervisor effectively becomes an agent but they may return to the supervisor role if another agent joins the conference. 5) A supervisor should be able to kick a caller. 6) A supervisor can press a key to hear how many callers and agents are in the conference. 7) An agent should log in to the system so we can track their time. 8) An agent can kick a caller. 9) An agent can press a key to hear how many callers are in the conference. 10) A caller can exit the conference by pressing a key. All 3 applications can handle parts, which one do you think is the best fit to add to? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline [EMAIL PROTECTED]Fax: +1-760-731-3000 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
On Mon, 2003-10-06 at 20:47, Brian West wrote: Works fine on my 7960 with 5.3 firmware. bkw Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. Hello, I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference. When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible. Thanks Babak -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
On Tue, 2003-10-07 at 12:09, Brian West wrote: Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I dont know how to park a call with the 7960. BTW, heres an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. bkw On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote: On Tue, 2003-10-07 at 12:09, Brian West wrote: Im having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I dont know how to park a call with the 7960. BTW, heres an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
Brian, Would you be kind enough to give me a brief overview of why it doesnt work. I also appreciate the work aorund. This is something I will have to educate my soon to be users on. We do a lot of conferencing of calls as a matter of facilitating clients' immediate needs. For now I will try parking one or more of the calls and conferencing via calling the park extension. I already have a meetme room setup, but it's not quite as convenient as asking someone to hangon while you get the other parties on the line to work out an issue. Especially since it is our policy to authenticate all meetmes. Thanks for everyone's response to this issue. Babak Brian West wrote: I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. bkw On Tue, 7 Oct 2003, Juan J. Sierralta P. wrote: On Tue, 2003-10-07 at 12:09, Brian West wrote: I´m having a similar problem with my 7960 when I receive two incoming calls I cannot join them. ya you can't join them. That sucks.. but you can park one call, go back to call number 1. Press conf. Dial the parking orbit.. then press join! How ? I don´t know how to park a call with the 7960. BTW, here´s an URL which may be related with the 3-way bug on 7960s. http://paf.se/inoc-dba/17.html -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
On Tue, 2003-10-07 at 15:38, Brian West wrote: I dont see it as a bug.. I see why it don't work.. and why people think it should. Enable # transfers.. and setup call parking to get around this. Also if you conf very much look at app_meetme. I did it, problem that I have now is the dialplan on the Cisco phone, as soon as I push # it dial without any number :( Im trying to get some info on dialplan.xml if somebody has an example to avoid the effect of the # I will appreciate it. Thanks! -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing Calls on Cisco 7940
Hello, I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference. When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible. Thanks Babak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing Calls on Cisco 7940
Works fine on my 7960 with 5.3 firmware. bkw On Mon, 6 Oct 2003, Babak Pasdar wrote: Hello, I am trying to conference two or more calls on a Cisco 7940 phone. When I have one inbound call and one outbound (I initiate the second call by pressing conference) I get the join button at the bottom of the screen and I can conference. When I initiate both calls or I receive both calls I dont get the join button. As a side question what would represent the hook flash on a Cisco 7940 or is this capability not possible. Thanks Babak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conferencing : authentication
The key functionality isn't in Asterisk right now, but someone has notified us that they've written it. We're waiting to get the patch and disclaimer from them to incorporate the changes. In the mean time you can use Authenticate application: exten = 8600,1,Authenticate(4321) exten = 8600,2,Meetme,1234 Mark On Mon, 2 Jun 2003, Rahul Gupta wrote: Hello ! I made some changes in the extension.conf to make * ask for a key when a person enters a conference room. But it is not asking for any key and takes the user directly to the conference room. I am using sip softphones as end points connected to * for conferencing. Any pointers for the solution ? relevent line of extension.conf looks like : exten = 8600,1,Meetme,1234|p|4321 thanks rahul __ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo. http://search.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferencing : authentication
Hello ! I made some changes in the extension.conf to make * ask for a key when a person enters a conference room. But it is not asking for any key and takes the user directly to the conference room. I am using sip softphones as end points connected to sip for conferencing. Any pointers for the solution ? relevent line of extension.conf looks like : exten = 8600,1,Meetme,1234|p|4321 thanks rahul __ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo. http://search.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
