Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
This can be done like this:
;user extensions
exten= 1,1,Dial(SIP/U1,,Tt)
exten= 2,1,Dial(SIP/U2,,Tt)
exten= 3,1,Dial(SIP/U2,,Tt)
;secretary extensions
exten= 4,1,Dial(SIP/Secretary,Tt)
the Tt option in dialplan lets the secretary to transfer the user
;conference extensions
exten= 123,1,Meetme(${EXTEN})
exten= 234,1,Meetme(${EXTEN})
your secratary dials any user who she wants to join the boss in a conference
room. after user answeres your secratary presses # button (or transfer
combination keys defined in features.conf) she will hear a transfer message
followed by a dialtone, here she has to dial conference room no where she
wants to throw the user to join the conference like 123, or 234.
Hope its helpfull
On 3/20/07, Angel Heart [EMAIL PROTECTED] wrote:
Hi Yehavi,
Yes, this can be done. We are currently using this features. The
Secretaries making the calls to who ever her Boss wants to join the
conference she then just transfer the calls into the conference room. You
can even annouce the name of the newly arrived calls in the conference.
Like; Mr. Mateevitsi join the conference or Mr. Mateevitsi leaved the
conference if one's leave the conference. I had created one coference room
for every department.
Regards.
Angel
*Victor Mateevitsi [EMAIL PROTECTED]* wrote:
Or, you can just transfer the calls into the conference room.
On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Yehavi Bourvine +972-8-9489444 wrote:
Why not use the MeetMe feature of asterisk?
I need the person who initiated the conference call to call the others
and join
them by herself. If I understand correctly, with the MeetMe you have
to
initialize the conference and then people dial by themselves into it.
This
won't be acceptable by the secretaries here...
Yehavi,
Can you make a script that uses call files to get everyone into the
conference?
--
Warm Regards,
Lee
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Software Engineer
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Re: [asterisk-users] Conference server (or how to make a call with more than 3 users)
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? 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] Conference server (or how to make a call with more than 3 u
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: ___ --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] Conference server (or how to make a call withmore than 3 u
Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: 19. marts 2007 09:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Conference server (or how to make a call withmore than 3 u On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: Hello, On most SIP phones a conference call is done on the phone and is limited to 3 participants. Polycom phones has a configuration option to use a conference server instead of the internal conferencing feature. I guess I need some conference server; any experience with such a server which can interact with Asterisk? Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Conference server (or how to make a call withmore than 3 u
Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I have a few Snom phones here - people do not like them... Thanks, __Yehavi: ___ --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] Conference server (or how to make a call withmore than 3 u
Jon Schøpzinsky wrote: Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I don't think this is true. The Snoms do not have enough CPU power for 12 people in a conference *on the phone*. And I doubt that it works for 6. Does it? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Conference server (or how to make a call withmorethan 3 u
With 6 people it works, we have tried it. The 12 people is, as I said, only in theory, because, as you said, the CPU is probably not powerful enough. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: 19. marts 2007 09:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Conference server (or how to make a call withmorethan 3 u Jon Schøpzinsky wrote: Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. I don't think this is true. The Snoms do not have enough CPU power for 12 people in a conference *on the phone*. And I doubt that it works for 6. Does it? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
Yehavi Bourvine +972-8-9489444 wrote: Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee ___ --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] Conference server (or how to make a call with more than 3 u
Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee Possible, but looks too much cumbersome... However, that's a nice idea. Thanks! __Yehavi: ___ --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] Conference server (or how to make a call with more than 3 u
Or, you can just transfer the calls into the conference room. On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote: Yehavi Bourvine +972-8-9489444 wrote: Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference server (or how to make a call with more than 3 u
Hi Yehavi, Yes, this can be done. We are currently using this features. The Secretaries making the calls to who ever her Boss wants to join the conference she then just transfer the calls into the conference room. You can even annouce the name of the newly arrived calls in the conference. Like; Mr. Mateevitsi join the conference or Mr. Mateevitsi leaved the conference if one's leave the conference. I had created one coference room for every department. Regards. Angel Victor Mateevitsi [EMAIL PROTECTED] wrote: Or, you can just transfer the calls into the conference room. On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote: Yehavi Bourvine +972-8-9489444 wrote: Why not use the MeetMe feature of asterisk? I need the person who initiated the conference call to call the others and join them by herself. If I understand correctly, with the MeetMe you have to initialize the conference and then people dial by themselves into it. This won't be acceptable by the secretaries here... Yehavi, Can you make a script that uses call files to get everyone into the conference? -- Warm Regards, Lee ___ --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 - Don't get soaked. Take a quick peek at the forecast with theYahoo! Search weather shortcut.___ --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] Conference Server
Hi, First, open your eyes and read the messages. Second use google. Google is there for just such a problem. Visit this url and marvel at how easy it is to ask google a question. http://tinyurl.com/2ajso I responded to a message not but half a month ago to tell the person to do the same thing. Install the kernel source. Sorry to bother you with my questions. I've installed the kernel source code. Compiled it and it runned smoothly. I have installed linux-2.4.22-1.2115.nptl and linux-2.6.6 sources. But, when I try to compile zaptelrtc with 2.6.6 I get the following error message: [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall In file included from /usr/src/linux/include/asm/processor.h:18, from /usr/src/linux/include/asm/thread_info.h:16, from /usr/src/linux/include/linux/thread_info.h:21, from /usr/src/linux/include/linux/spinlock.h:12, from /usr/src/linux/include/linux/capability.h:45, from /usr/src/linux/include/linux/sched.h:7, from /usr/src/linux/include/linux/module.h:10, from zaprtc.c:60: /usr/src/linux/include/asm/system.h: En la función `__set_64bit_var': /usr/src/linux/include/asm/system.h:193: aviso: dereferencing type-punned pointer will break strict-aliasing rules /usr/src/linux/include/asm/system.h:193: aviso: dereferencing type-punned pointer will break strict-aliasing rules En el fichero incluído de /usr/src/linux/include/asm/smp.h:18, de /usr/src/linux/include/linux/smp.h:17, de /usr/src/linux/include/linux/sched.h:23, de /usr/src/linux/include/linux/module.h:10, de zaprtc.c:60: /usr/src/linux/include/asm/mpspec.h:6:25: mach_mpspec.h: No existe el fichero o el directorio In file included from /usr/src/linux/include/asm/smp.h:18, from /usr/src/linux/include/linux/smp.h:17, from /usr/src/linux/include/linux/sched.h:23, from /usr/src/linux/include/linux/module.h:10, from zaprtc.c:60: /usr/src/linux/include/asm/mpspec.h: En el nivel principal: /usr/src/linux/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:9: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:10: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:12: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:19: error: `MAX_APICS' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:20: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:20: error: conflicting types for `mp_bus_id_to_type' /usr/src/linux/include/asm/mpspec.h:8: error: previous declaration of `mp_bus_id_to_type' /usr/src/linux/include/asm/mpspec.h:22: error: `MAX_IRQ_SOURCES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:24: error: `MAX_MP_BUSSES' undeclared here (not in a function) /usr/src/linux/include/asm/mpspec.h:24: error: conflicting types for `mp_bus_id_to_pci_bus' /usr/src/linux/include/asm/mpspec.h:12: error: previous declaration of `mp_bus_id_to_pci_bus' /usr/src/linux/include/asm/mpspec.h:43: error: `MAX_APICS' undeclared here (not in a function) and the error output continues... and when I try to compile it with 2.4.22-1.2115.nptl, I get this error output [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall -DMODVERSIONS -include /usr/src/linux/include/linux/modversions.h En el fichero incluído de /usr/src/linux/include/linux/spinlock.h:56, de /usr/src/linux/include/linux/module.h:11, de zaprtc.c:60: /usr/src/linux/include/asm/spinlock.h:9:12: sufijo b7d4074 inválido en la constante entera In file included from /usr/src/linux/include/linux/spinlock.h:56, from /usr/src/linux/include/linux/module.h:11, from zaprtc.c:60: /usr/src/linux/include/asm/spinlock.h:9: error: error sintáctico before numeric constant /usr/src/linux/include/asm/spinlock.h:10: error: `printk_R_ver_str' declared as function returning a function In file included from /usr/src/linux/include/linux/prefetch.h:13, from /usr/src/linux/include/linux/list.h:6, from /usr/src/linux/include/linux/module.h:12, from zaprtc.c:60: /usr/src/linux/include/asm/processor.h:60: aviso: nombres de parámetros (sin tipos) en la declaración de la función /usr/src/linux/include/asm/processor.h:60: error: field `loops_per_jiffy_R_ver_str' declared as a function and the error output continues... I've
Re: [Asterisk-Users] Conference Server
HI there, Thanks everybody for all the answers. I took a look at the asterisk timer ztdummy page (http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy) Unfortunaly, my PC has the USB OHCI module. So, I downloaded the zaprtc module from http://www.junghanns.net/asterisk/. I tried to do make, and got the following error message: [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall En el fichero includo de /usr/include/linux/module.h:20, de zaprtc.c:60: /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h:2:2: #error but rather headers from an appropriate kernel-source package. /usr/include/linux/modversions.h:3:2: #error Change -I/usr/src/linux/include (or similar) to /usr/include/linux/modversions.h:4:2: #error -I/lib/modules/$(uname -r)/build/include /usr/include/linux/modversions.h:5:2: #error to build against the currently-running kernel. In file included from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timex.h:56: error: error sintctico before and In file included from /usr/include/linux/timex.h:126, from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/asm/timex.h:33: error: error sintctico before cacheflush_time /usr/include/asm/timex.h:35: error: error sintctico before get_cycles In file included from /usr/include/linux/sched.h:14, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timex.h:147: error: field `time' has incomplete type En el fichero includo de /usr/include/linux/bitops.h:69, de /usr/include/asm/system.h:7, de /usr/include/linux/sched.h:16, de /usr/include/linux/mm.h:4, de /usr/include/linux/locks.h:5, de /usr/include/linux/devfs_fs_kernel.h:6, de /usr/include/linux/miscdevice.h:4, de zaprtc.c:63: /usr/include/asm/bitops.h:327:2: aviso: #warning This includefile is not available on all architectures. /usr/include/asm/bitops.h:328:2: aviso: #warning Using kernel headers in userspace: atomicity not guaranteed In file included from /usr/include/linux/signal.h:4, from /usr/include/linux/sched.h:25, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/asm/signal.h:107: error: error sintctico before sigset_t /usr/include/asm/signal.h:110: error: error sintctico before '}' token In file included from /usr/include/linux/sched.h:81, from /usr/include/linux/mm.h:4, from /usr/include/linux/locks.h:5, from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/timer.h:45: error: error sintctico before spinlock_t /usr/include/linux/timer.h:53: error: error sintctico before '}' token /usr/include/linux/timer.h:67: error: error sintctico before tvec_base_t /usr/include/linux/timer.h:101: error: error sintctico before tvec_bases /usr/include/linux/timer.h: En la funcin `init_timer': /usr/include/linux/timer.h:105: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h:105: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h:106: error: dereferencing pointer to incomplete type /usr/include/linux/timer.h: En la funcin `timer_pending': /usr/include/linux/timer.h:121: error: dereferencing pointer to incomplete type En el fichero includo de /usr/include/linux/devfs_fs_kernel.h:6, de /usr/include/linux/miscdevice.h:4, de zaprtc.c:63: /usr/include/linux/locks.h:8:27: linux/pagemap.h: No existe el fichero o el directorio In file included from /usr/include/linux/devfs_fs_kernel.h:6, from /usr/include/linux/miscdevice.h:4, from zaprtc.c:63: /usr/include/linux/locks.h: En la funcin `wait_on_buffer': /usr/include/linux/locks.h:19: error:
Re: [Asterisk-Users] Conference Server
On Fri, 2004-05-28 at 10:53, pesb wrote: HI there, Thanks everybody for all the answers. I took a look at the asterisk timer ztdummy page (http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy) Unfortunaly, my PC has the USB OHCI module. So, I downloaded the zaprtc module from http://www.junghanns.net/asterisk/. I tried to do make, and got the following error message: [EMAIL PROTECTED] zaptelrtc]# make cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/usr/src/linux/include -Wall En el fichero incluído de /usr/include/linux/module.h:20, de zaprtc.c:60: /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, How can I install zaprtc on my PC. I have a PIV Fedora Core 1 with a 2.4.22-1.2115.nptl kernel? First, open your eyes and read the messages. Second use google. Google is there for just such a problem. Visit this url and marvel at how easy it is to ask google a question. http://tinyurl.com/2ajso I responded to a message not but half a month ago to tell the person to do the same thing. Install the kernel source. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conference Server
I think if you use ztdummy that is all that is required. Un comment in the zaptel Makefile and recompile. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 27 May 2004 16:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Server Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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
Re: [Asterisk-Users] Conference Server
ztdummy will suffice. A Zaptel interface is used as a timing device for the conference. On Thu, 2004-05-27 at 11:58, pesb wrote: Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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
Re: [Asterisk-Users] Conference Server
Hi, take a look at zaprtc (which generates the zaptel timing out of your pc's realtime clock) or ztdummy (which uses an usb-uhci controller to generate the timing). best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Do, 2004-05-27 um 17.58 schrieb pesb: Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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
Re: [Asterisk-Users] Conference Server
pesb wrote: Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. No...go here: http://store.yahoo.com/asteriskpbx/wildcardx100p.html Anything less is a hack and will cause problems, especially at higher levels of scale. Jeremy McNamara ___ 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] Conference Server
If you have usb hardware installed, you can use the ztdummy driver (part of the zaptel bits), and you don't need usb hardware if you're using a 2.6 kernel IIRC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of pesb Sent: 27 May 2004 16:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Server Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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 This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] Conference Server
pesb [EMAIL PROTECTED] wrote: I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. There are various companies who will sell you a X100P card for around the US$15 mark. Examples can be found at www.govarion.com, www.digitnetworks.com and www.goods2world.com. Failing that, you could install the ztdummy kernel module, as explained here: www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] Conference Server
There is a dummy driver calles ztdummy which can loaded to have those functionality. Have a look onto http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy Regards Julian Pawlowski ___ 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] Conference server
Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP kernel and have about 30 channels in conference. Here's the bug listing: http://bugs.digium.com/bug_view_page.php?bug_id=963 MATT--- -Original Message- From: Paulo Mannheimer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference server Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ 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
RE: [Asterisk-Users] Conference server
This seems to only apply to non-zap channels participating in the conference, incidently. On Fri, 6 Feb 2004, mattf wrote: Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP kernel and have about 30 channels in conference. Here's the bug listing: http://bugs.digium.com/bug_view_page.php?bug_id=963 MATT--- -Original Message- From: Paulo Mannheimer [mailto:[EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:20 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference server Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM ___ 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 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
