[Asterisk-Users] Re: Meetme
In article <[EMAIL PROTECTED]>, Tim Sailer <[EMAIL PROTECTED]> wrote: > OK, maybe I need more coffee. Or less. Either way, I'm stumped. > > I have a Meetme conference room configured. Meetme(|M|) to enable > the MOH. When you are the first one to go into the conf, you get the > announcement that you are the only one, and then a *male* voice gives > a little talk about 'Why are we putting you on hold?'. Where does that > come from and how do I get rid of it? It's not any of the sound files > that I can see/hear, and running 'asterisk -vvvr' doesn't show any > file being played... Have a look in /var/lib/asterisk/mohmp3 - there is a sample file in there with what you describe. Delete or move that file, and put your own MP3s in that directory. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: meetme
In article <[EMAIL PROTECTED]>, Travis Conway <[EMAIL PROTECTED]> wrote: > Aug 6 13:48:56 WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No application > 'MeetMe' for extension (from-sip, 9000, 4) Check that you have the file /usr/lib/asterisk/modules/app_meetme.so and that /etc/asterisk/modules.conf has a [modules] section with either: * A line saying "autoload=yes" and NO line saying "noload => app_meetme.so" * A line saying "load => app_meetme.so" Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: Meetme
Steve Kann <[EMAIL PROTECTED]> writes: > ([app_conference is] located in iaxclient CVS at iaxclient.sf.net). Not any more, it isn't. :-( Anyone know if it's still available somewhere? -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Re: Meetme
Michael Bielicki <[EMAIL PROTECTED]> writes: > http://cvs.sourceforge.net/viewcvs.py/iaxclient/app_conference/ Oops -- my bad. I was looking inside .../iaxclient/iaxclient/... Thanks! -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Re: Meetme
Steve Kann <[EMAIL PROTECTED]> writes: > I don't think it's in the Wiki, and it's not really documented; Could you offer a very, very brief introduction? I've figured out, through trial and error, that it takes a call to Conference(somename) in an extension to create or join a conference, but I can't get anyone connected in any other state than "listener", and there is no sound. Am I missing a parameter, a configuration file, or what...? > We've talked about it a bit on iaxclient-devel mailing list; I searched the list, but the closest I came was Steven Sokol asking how to use app_conference, with no answer archived... ;-) -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Re: Meetme
Darren Wiebe <[EMAIL PROTECTED]> writes: > If you've got it running that means it built for you. Did it build out > of the box? I've tried changing the paths in the Makefile to the > correct ones but it still dies with the following error. There were a few changes, some of which seem to be needed because things have changed in Asterisk itself. Here's what I did: Index: Makefile === RCS file: /cvsroot/iaxclient/app_conference/Makefile,v retrieving revision 1.7 diff -c -r1.7 Makefile *** Makefile7 Jul 2004 13:39:41 - 1.7 --- Makefile23 Sep 2004 17:19:43 - *** *** 17,32 # app_conference defines which can be passed on the command-line # ! INSTALL_PREFIX := /opt/horizon INSTALL_MODULES_DIR := $(INSTALL_PREFIX)/lib/asterisk/modules ! ASTERISK_INCLUDE_DIR := $(HOME)/local/asterisk/asterisk/include # turn app_conference debugging on or off ( 0 == OFF, 1 == ON ) APP_CONFERENCE_DEBUG := 1 # 0 = OFF 1 = astdsp 2 = speex ! SILDET := 2 # # app_conference objects to build --- 17,32 # app_conference defines which can be passed on the command-line # ! INSTALL_PREFIX := /usr INSTALL_MODULES_DIR := $(INSTALL_PREFIX)/lib/asterisk/modules ! ASTERISK_INCLUDE_DIR := /usr/include/asterisk # turn app_conference debugging on or off ( 0 == OFF, 1 == ON ) APP_CONFERENCE_DEBUG := 1 # 0 = OFF 1 = astdsp 2 = speex ! SILDET := 0 # # app_conference objects to build *** *** 44,50 CC = gcc INCLUDE = -I$(ASTERISK_INCLUDE_DIR) ! LIBS = -ldl -lpthread -lm DEBUG := -g CFLAGS = -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE --- 44,50 CC = gcc INCLUDE = -I$(ASTERISK_INCLUDE_DIR) ! LIBS = -lpthread -lm DEBUG := -g CFLAGS = -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations $(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE *** *** 53,64 # PERF: below is 10% faster than -O2 or -O3 alone. #CFLAGS += -O3 -ffast-math -funroll-loops # below is another 5% faster or so. ! CFLAGS += -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant ! CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic # adding -msse -mfpmath=sse has little effect. #CFLAGS += -O3 -msse -mfpmath=sse #CFLAGS += $(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi) ! CFLAGS += $(shell if uname -m | grep -q ppc; then echo "-fsigned-char"; fi) CFLAGS += -DCRYPTO ifeq ($(APP_CONFERENCE_DEBUG), 1) --- 53,65 # PERF: below is 10% faster than -O2 or -O3 alone. #CFLAGS += -O3 -ffast-math -funroll-loops # below is another 5% faster or so. ! CFLAGS += -O3 -ffast-math -funroll-all-loops -fsingle-precision-constant ! #CFLAGS += -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant ! #CFLAGS += -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic # adding -msse -mfpmath=sse has little effect. #CFLAGS += -O3 -msse -mfpmath=sse #CFLAGS += $(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi) ! #CFLAGS += $(shell if uname -m | grep -q ppc; then echo "-fsigned-char"; fi) CFLAGS += -DCRYPTO ifeq ($(APP_CONFERENCE_DEBUG), 1) *** *** 102,110 install: all for x in $(SHAREDOS); do $(INSTALL) -m 755 $$x $(INSTALL_MODULES_DIR) ; done ! /var/horizon/mojo/lib/horizoncmd restart asterisk # config: all # cp conf.conf /etc/asterisk/ ! --- 103,111 install: all for x in $(SHAREDOS); do $(INSTALL) -m 755 $$x $(INSTALL_MODULES_DIR) ; done ! # /var/horizon/mojo/lib/horizoncmd restart asterisk # config: all # cp conf.conf /etc/asterisk/ ! Index: conference.c === RCS file: /cvsroot/iaxclient/app_conference/conference.c,v retrieving revision 1.4 diff -c -r1.4 conference.c *** conference.c7 Jul 2004 13:39:41 - 1.4 --- conference.c23 Sep 2004 17:19:44 - *** *** 26,35 static struct ast_conference *conflist = NULL ; // mutex for synchronizing access to conflist ! static ast_mutex_t conflist_lock = AST_MUTEX_INITIALIZER ; // mutex for synchronizing calls to start_conference() and remove_conf() ! static ast_mutex_t start_stop_conf_lock = AST_MUTEX_INITIALIZER ; static int conference_count = 0 ; --- 26,35 static struct ast_conference *conflist = NULL ; // mutex for synchronizing access to conflist ! AST_MUTEX_DEFINE_STATIC( conflist_lock ) ; // mutex for synchronizing calls to start_conference() and remove_conf() ! AST_MUTEX_DEFINE_STATIC( start_stop_conf_lock ) ; static int conference_count = 0 ; *** *** 604,610
[Asterisk-Users] Re: Meetme
Steve Kann <[EMAIL PROTECTED]> writes: > This was what I wrote on the iaxclient list previously: Cool, Steve -- thanks a lot! Conference() works great for me now. :-) I've extended the description for "show application" thus: static char *descrip = " Conference([/[/[/probstart[/probcont):\n" "Creates or joins a telephone conference. There is no configuration file;\n" "everything is controlled through parameters to the invocation from an\n" "extension context.\n" " : an alphanumeric string identifying the conference\n" " : a concatenation of flag characters chosen from the following:\n" " M: user is a moderator, i.e. is allowed to speak\n" " L: user is a listener, i.e. may only listen in (default)\n" " T: user has a telephone, not an iaxclient; enable speex\n" " V: sets speex flag SPEEX_PREPROCESS_SET_VAD\n" " D: sets speex flag SPEEX_PREPROCESS_SET_DENOISE\n" " A: sets speex flag SPEEX_PREPROCESS_SET_AGC\n" " : not currently used\n" " : sets SPEEX_PREPROCESS_SET_PROB_START value\n" " : sets SPEEX_PREPROCESS_SET_PROB_CONTINUE value\n" "Returns 0 if the user exits with the '#' key, or -1 if the user hangs up.\n" ; -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Re: meetme
In article <[EMAIL PROTECTED]>, Giuseppe <[EMAIL PROTECTED]> wrote: > Hi, > when I try to use meetme I always hear this error message > "this is not a valid conference number, please try again", > but my configuration seems to be correct... Here it is: > > -- extensions.conf -- > exten => 6000,1,MeetMe(1234,ciMp) ; entra nella meetme room 1234 > > -- meetme.conf -- > [rooms] > conf => 1234 > > Does anyone has the same problem? Any idea? You need to use '|' as the separator instead of ','. I think your example above is trying to enter a conference called "1234,ciMp". Also, I always find is best to answer the line and wait a litle bit before calling MeetMe: exten => 6000,1,Answer exten => 6000,2,Wait(0.5) exten => 6000,3,MeetMe(1234|ciMp) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme
On Thu, Mar 04, 2004 at 02:53:20PM -, Tony Mountifield wrote: > In article <[EMAIL PROTECTED]>, > Tim Sailer <[EMAIL PROTECTED]> wrote: > > OK, maybe I need more coffee. Or less. Either way, I'm stumped. > > > > I have a Meetme conference room configured. Meetme(|M|) to enable > > the MOH. When you are the first one to go into the conf, you get the > > announcement that you are the only one, and then a *male* voice gives > > a little talk about 'Why are we putting you on hold?'. Where does that > > come from and how do I get rid of it? It's not any of the sound files > > that I can see/hear, and running 'asterisk -vvvr' doesn't show any > > file being played... > > Have a look in /var/lib/asterisk/mohmp3 - there is a sample file in > there with what you describe. Delete or move that file, and put your > own MP3s in that directory. OK, that's strange. I took out the sample file a long time ago! I guess an upgrade put it back... sheesh. Gotta be careful about that. That's strange. I must have hit the timing *perfectly*... the 3 times I tried this, that mp3 played as soon as I entered the room, and it's supposed to be a random play... I think it's time I played lotto this week! :) Tim PS: Thanks. That was driving me nuts! -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 << >< ___ 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] Re: Meetme question
If you know what you want the conf room number to be, then set that up in
meetme.conf.
You would have to write your own IVR though, and use Authenticate() with the
PIN kept in the DB. Its a hack but it would do what you want:
exten => _X,1,Playback("conf-getconfno")
exten => _9XX,1,Authenticate(${DB(conf/${EXTEN})})
exten => _9XX,1,MeetMe(${EXTEN},s,)
I don't know any other way to have a PIN number that isn't statically
defined in the meetme.conf file or created on the fly with the room useing
the D option, if you know another way let me know.
--
Justin
Hi,
I'm experimenting with the Meetme feature of Asterisk 1.2,
exten => 2095,1,MeetMe(|Ds)
This almost gives me what I want, where each employee can create their own
on-the-fly conferences with a personal Conference Number and PIN. However, as
the PIN is actually set by the first callee, then its subject to problems
(first callee might enter the wrong PIN, and then no-one else can join).
What I really want is something that covers the below:
- One call-in number
- Employees get their own unique conference # (this could be their own
extension), and can set a public PIN that only they can change.
- I don't really want a www-based system, as most of my users are usually
mobile, and might not have access to the corporate intranet.
Thanks,
Adrian Marsh
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[asterisk-users] Re: meetme sounds
In article <[EMAIL PROTECTED]>, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > I am playing around with dynamic meetme conferences, and wanted to have > one person constantly in the conference, with calls "popping in and out". > > Is there an option / any way of playing enter / leave sounds to the > person who created the conference only, and not the people leaving / > joining ? Not without custom modifications to the code of app_meetme itself. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme Announcement
In article <[EMAIL PROTECTED]>, Mohamed Farid <[EMAIL PROTECTED]> wrote: > > Dear All : > How can I enable the announcement Feature of Meet-me rooms ? > So that when I enter the conference room , the system ask me about my > name ,, then announce all the existing people in the room about my > entrance .. > Also when I go out of the conference - an announce should be played to > all the remaining members in the conference saying that I am out ... Pass the 'i' option to MeetMe in your dialplan. Note that this option is only available in CVS HEAD, not in STABLE nor in any of the 1.0.x releases. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] RE: MeetMe Improvement
is there any option of inviting some one to conference, I mean, I press * for menu, then system asks me to invite some one dial 1, and then asks me to dial the extension of that person, and then call is placed to invite that person to conference. Thank you Atif Sent via the WebMail system at convergence.com.pk ___ 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] Re: meetme
I just downloaded the new stuff form CVS and compiled it, but cannot find the meetme so file. What gives? -- Travis Conway [EMAIL PROTECTED] FWD: 414668 +1 334 220-7519 (T-Mobile) - Original Message - From: "Tony Mountifield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 06, 2004 2:26 PM Subject: [Asterisk-Users] Re: meetme > In article <[EMAIL PROTECTED]>, > Travis Conway <[EMAIL PROTECTED]> wrote: > > Aug 6 13:48:56 WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No application > > 'MeetMe' for extension (from-sip, 9000, 4) > > Check that you have the file /usr/lib/asterisk/modules/app_meetme.so > and that /etc/asterisk/modules.conf has a [modules] section with either: > > * A line saying "autoload=yes" and NO line saying "noload => app_meetme.so" > > * A line saying "load => app_meetme.so" > > Cheers > Tony > -- > Tony Mountifield > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > ___ > 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] Re: meetme application
In article <[EMAIL PROTECTED]>, pesb <[EMAIL PROTECTED]> wrote: > Hi there, > I know this question is kind of stupid. But, I don't know anywhere > else to ask. I've received some answers when I asked about the need of having > a zaptel interface to make the meetme application work, that said that it was > better to have a real hardware then the zaptelrtc software modules. > So, my question is, would any of the following cards work with the meetme > application? > > http://www.goods2world.com/product_info.php?cPath=25&products_id=53 > http://store.yahoo.com/asteriskpbx/wildcardx100p.html > http://www.digitnetworks.com/store/product_info.php?products_id=28&osCsid=a77255f932eb7a247387647e1da6dbb3 > http://www.digium.com/index.php?menu=wildcard_x100p > http://69.212.249.116/product_info.php?products_id=30&osCsid=d4225af643f710ece118ad93a64347f9 They all will. However, by buying from store.yahoo.com or www.digium.com you will be helping to support Digium, who gave Asterisk PBX to the community. This is a Good Thing (TM). The other three are OEM copies of the same card, but do not benefit Digium at all. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: Meetme
http://cvs.sourceforge.net/viewcvs.py/iaxclient/app_conference/ -- Michael Bielicki ___ 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] Re: Meetme
On Sep 23, 2004, at 4:46 AM, Tom Ivar Helbekkmo wrote: Steve Kann <[EMAIL PROTECTED]> writes: ([app_conference is] located in iaxclient CVS at iaxclient.sf.net). Not any more, it isn't. :-( Anyone know if it's still available somewhere? Sure it is: http://sourceforge.net/cvs/?group_id=72851 http://cvs.sourceforge.net/viewcvs.py/iaxclient/app_conference/ -SteveK ___ 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] Re: Meetme
If you've got it running that means it built for you. Did it build out of the box? I've tried changing the paths in the Makefile to the correct ones but it still dies with the following error. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk-old -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/asterisk-old -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c: In function `create_conf': conference.c:607: warning: implicit declaration of function `__use_ast_pthread_create_instead__' make: *** [conference.o] Error 1 Darren Tom Ivar Helbekkmo wrote: Steve Kann <[EMAIL PROTECTED]> writes: I don't think it's in the Wiki, and it's not really documented; Could you offer a very, very brief introduction? I've figured out, through trial and error, that it takes a call to Conference(somename) in an extension to create or join a conference, but I can't get anyone connected in any other state than "listener", and there is no sound. Am I missing a parameter, a configuration file, or what...? We've talked about it a bit on iaxclient-devel mailing list; I searched the list, but the closest I came was Steven Sokol asking how to use app_conference, with no answer archived... ;-) -tih ___ 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] Re: Meetme
On Sep 23, 2004, at 8:10 AM, Tom Ivar Helbekkmo wrote: Steve Kann <[EMAIL PROTECTED]> writes: I don't think it's in the Wiki, and it's not really documented; Could you offer a very, very brief introduction? I've figured out, through trial and error, that it takes a call to Conference(somename) in an extension to create or join a conference, but I can't get anyone connected in any other state than "listener", and there is no sound. Am I missing a parameter, a configuration file, or what...? This was what I wrote on the iaxclient list previously: the param string that is passed to app_conference is of this form: - 'conference_id' can be alphanumberic, i.e. MyConference or 1234567890 - 'user_flags' are: // user type flags M -> moderator, i.e. user can speak L -> listener, i.e. user cannot speak S -> sip, same as moderator for now note: last flag in the string wins. so 'SLM' would make the user a moderator, and 'MSL' would make the user a listener. // speex flags V -> sets SPEEX_PREPROCESS_SET_VAD flag D -> sets SPEEX_PREPROCESS_SET_DENOISE flag A ->sets SPEEX_PREPROCESS_SET_AGC ( auto gain control ) flag T -> user is calling from a telephone. used to enable speex preprocessing, since iaxclient performs speex preprocessing on client side. - 'priority' is not currently used. - 'vad_prob_start' and 'vad_prob_continue' are optional and have DEFINE'd defaults. vad_prob_start -> sets SPEEX_PREPROCESS_SET_PROB_START value vad_prob_continue -> sets SPEEX_PREPROCESS_SET_PROB_CONTINUE value note: these only work with the libspeex included with app_conference. our patches to libspeex which support these flags are not yet in the published libspeex code. basically, there's no conf file, and you call it like this from extensions.conf: ; enter with VAD, as if it were a telephone call exten => _7,1,Playback(beep) exten => _7,2,Conference(TestConference/MTV/1); ; enter as a speaker, with no vad. exten => _8,1,Playback(beep) exten => _8,2,Conference(TestConference/M/1); ; enter as a listener. exten => _9,1,Playback(beep) exten => _9,2,Conference(TestConference/L/1); If you're at astricon, look for me! (I'm wearing a bright orange Digium shirt today). -SteveK ___ 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] Re: Meetme
Hi all, Is there any basic information available for app_conferense? Does it suport SIP and other codecs Any installation guide Thanks Umar -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tom Ivar Helbekkmo Sent: 24 September 2004 06:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Meetme Steve Kann <[EMAIL PROTECTED]> writes: > This was what I wrote on the iaxclient list previously: Cool, Steve -- thanks a lot! Conference() works great for me now. :-) I've extended the description for "show application" thus: static char *descrip = " Conference([/[/[/probstart[/probcont):\n" "Creates or joins a telephone conference. There is no configuration file;\n" "everything is controlled through parameters to the invocation from an\n" "extension context.\n" " : an alphanumeric string identifying the conference\n" " : a concatenation of flag characters chosen from the following:\n" " M: user is a moderator, i.e. is allowed to speak\n" " L: user is a listener, i.e. may only listen in (default)\n" " T: user has a telephone, not an iaxclient; enable speex\n" " V: sets speex flag SPEEX_PREPROCESS_SET_VAD\n" " D: sets speex flag SPEEX_PREPROCESS_SET_DENOISE\n" " A: sets speex flag SPEEX_PREPROCESS_SET_AGC\n" " : not currently used\n" " : sets SPEEX_PREPROCESS_SET_PROB_START value\n" " : sets SPEEX_PREPROCESS_SET_PROB_CONTINUE value\n" "Returns 0 if the user exits with the '#' key, or -1 if the user hangs up.\n" ; -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Re: MeetME Conferencing
In article <[EMAIL PROTECTED]>, Mike Clark <[EMAIL PROTECTED]> wrote: > Damon Estep wrote: > > > Can anyone point me to a sample or information on using MeetMe like this? > > > > Conference room is set up with 2 PINs, one for the moderator and one > > for the participants. > > > > Participants get music until the moderator joins (to avoid wild, > > un-moderated tangents). > > > > Call is ended and all participants are kicked out when the moderator > > leaves (or the moderator can kick everyone out via phone keypad). > > > > Asking too much, or simple stuff? > > > Latest version of Web-MeetMe will do this, but it is definitely of the > add-on variety. > > You can do it pure dial plan if you are willing to have a menu that says > "Press 1 to join as admin, Press 2 to join as participant". Or you just define two different extensions - one for the admin to dial and one for normal users. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: meetme public
On Thu, Jun 08, 2006 at 12:03:32AM +0100, Marco Mouta wrote:
>
Marco. i solve this creating adding the meetme extension in the default
context. this extension now is valid for any user.
>Hi,
>Please check you [general] section in sip.conf
>; If you need to answer unauthenticated calls, you should change this
>; next line to 'from-trunk', rather than 'from-sip-external'.
>; You'll know this is happening if when you call in you get a message
>; saying "The number you have dialed is not in service. Please check
>the
>; number and try again."
>context = from-sip-external ; Send unknown SIP callers to this context
>callerid = Unknown
>It could be happening that your public sip call is arriving @
>asterisk, and seems unknow, so it is sent to from-sip-external
>context.
>In your extensions.conf look for section called [from-sip-external],
>there you need to paste your code to route the call to your meetme
>room.
>Hope it helps,
>Best regards,
>Marco Mouta
>Ps. Please give me some feeback if it solved.
>
>On 6/7/06, Pablo Allietti <[EMAIL PROTECTED]> wrote:
>
> hi all i have an asterisk working and i need to add a mettme public
> service.
> for example i need to download a soft (sjphone) and without any
> configuration call to [EMAIL PROTECTED] (meetme) and
> join a conference but when i do that i
> received an error saying nomber do not exist. but if i call a
> extension
> is work propperly.
> in the extensions.conf have
> exten => 411,1,Answer
> exten => 411,2,Wait(1)
> exten =>
> 411,3,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
> exten => 411,4,Monitor(wav,${TIMESTAMP},m)
> exten => 411,5,Meetme(4001,qM)
> exten => 411,6,Hangup
> 4001 is the room number
> in the mmetme conf have
> conf => 4001
> any comments?
> --
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>Com os melhores cumprimentos,
>Marco Mouta
>
> References
>
>1. mailto:[EMAIL PROTECTED]
>2. mailto:[EMAIL PROTECTED]
>3. http://Easynews.com/
>4. http://lists.digium.com/mailman/listinfo/asterisk-users
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.-
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Re: [Asterisk-Users] Re: meetme
Hi Tony, thanks for your answer! I tryed doing so, but I still get that error, sorry. Giuseppe - Tony Mountifield ha scritto: In article <[EMAIL PROTECTED]>, Giuseppe <[EMAIL PROTECTED]> wrote: Hi, when I try to use meetme I always hear this error message "this is not a valid conference number, please try again", but my configuration seems to be correct... Here it is: -- extensions.conf -- exten => 6000,1,MeetMe(1234,ciMp) ; entra nella meetme room 1234 -- meetme.conf -- [rooms] conf => 1234 Does anyone has the same problem? Any idea? You need to use '|' as the separator instead of ','. I think your example above is trying to enter a conference called "1234,ciMp". Also, I always find is best to answer the line and wait a litle bit before calling MeetMe: exten => 6000,1,Answer exten => 6000,2,Wait(0.5) exten => 6000,3,MeetMe(1234|ciMp) Cheers Tony ___ --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] Re: MeetMe Video option
That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/"to do" list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes. I was hoping to look into someday but I'm swamped with 1000 other things of higher priority. I have been thinking though, of some ways it could be supported, starting with the simplest and easiest: 1. First, if only 2 of the phones in the conference are video phones, allow them to exchange their video with each other, while having all of the audio streams conferenced as usual. 2a. The next step could be having each videophone "rotate" which stream it was showing for a few seconds (20 seconds maybe?). i.e. you could have 3 video calls mixed with several audio-only calls. Initially video call #1 would show #2's image, #2 would show #3's image, #3 would show #1's image for a few seconds, then rotate them by 1. Of course you don't need to show your own! :) Actually, ours has a picture-in-picutre in the corner so you can see yourself all the time anyway. 2b. The other option instead of time-rotating the images would be to try to show the image of whoever was talking. That kind of sounds like a pain to me, but maybe it's doable. 3. The really fancy thing would be to have Asterisk decode all of the video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them and send them to each client. That REALLY sounds like a pain to me, but again, maybe it's doable. Right now I'd be pretty happy with 2a though. - Matt Message: 3 From: "Regovich, Timothy" <[EMAIL PROTECTED]> To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> Date: Fri, 30 Jan 2004 13:07:46 -0500 Subject: [Asterisk-Users] MeetMe Video option Reply-To: [EMAIL PROTECTED] Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim ___ 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] Re: MeetMe Video option
No, there is no video output once the call goes into a meetme room. What I was talking about is a case where you have a regular video call between 2 video phones, then you try to send them to a conference room. The audio still works but the softphone's (Linphone in our case) behavior is to just freeze the video with the last image it received. I should mention one caveat to my previous suggestion (about just passing through the video with 2 phones or "rotating" through the images) ; I was assuming that all video calls were using the same format. In our case that would be true for the time being. That should be easy to do, just direct the rtp packets to the desired client. If they were using different video formats, you'd have to translate between them. From: "Regovich, Timothy" <[EMAIL PROTECTED]> To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Re: MeetMe Video option Date: Fri, 30 Jan 2004 16:02:55 -0500 Reply-To: [EMAIL PROTECTED] So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? ___ 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] Re: Meetme define context
In article <[EMAIL PROTECTED]>, Chris Blunt <[EMAIL PROTECTED]> wrote: > > I'm still having trouble trying to figure out if it is possible to define > (in the dial plan) a context for meetme? Just have a look in apps/app_meetme.c in the functions find_conf() and conf_exec(). You can see in there that "rooms" is hard-coded. > I'm using 1.4.4 with dialplan logic of: > > exten => 123,1,Meetme(,Msa,) > > This defaults to conferences defined within the rooms context of meetme.conf > > Is it possible to specify another context as with voicemail? [rooms] isn't a "context" in meetme, it's just a name for the section of the conf file where all the rooms are defined. > Or can any one think of another way to do this, my ultimate goal is to have > only certain conferences available to certain extension numbers. > > For example, call extension 123 have access to conference numbers > ,1112,1113 call extension 124 have access to conferences 1114,1115 etc. You would have to do your own logic in the dialplan instead of letting Meetme prompt for the conference number. Then you can check validity, and pass the conference number as the first arg to Meetme. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe and ChannelRedirect
In article <[EMAIL PROTECTED]>, Rafael Vidal Aroca <[EMAIL PROTECTED]> wrote: > i'm trying to implement the following scenario: > > - A user calls number 700 > - Asterisk then dials to extensions 100, 200, 300, 400 and 500 > - And then bridges all calls to a conference room > > I tried to use MeetMe and ChannelRedirect, but seems that after > channel redirect nothing more is executed. So, this seem to work for the > caller and first called, but the others stay outside. > > Could anyone help or give me a hint? The way I did this kind of thing was like this: 1. Extension 700 calls an AGI script which generates a .call file in /var/spool/asterisk/outgoing for each of the calls to the other extensions. Extension 700 then drops into the Meetme room to wait for the others. 2. Each call file specifies a Local channel to make the call to the extension, and uses the Context, Extension and Priority fields to direct the answered call into the Meetme room. If any of the calls to the other extensions fails (e.g. busy), you don't get any notification of that. If you want such notification, you will need to get a lot more complex, probably involving a controlling process using the Manager API. Hope this helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] RE: Meetme with video???
I wonder if there is an application available, what would allow me to have a conference call (meetme) with video. Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this functionality: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+video+conferencing You can add to this bounty, if you want. I'm trying to convince the money people at my company that we should add $500 to this. BTW: Is anybody working on this? ___ 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] Re: meetme - conf-invalid
In article <[EMAIL PROTECTED]>, Kevin Bockman <[EMAIL PROTECTED]> wrote: > > Yes, meetme requires a clock source. You could try ztdummy. I tried > > using an FXO card as a clock source and observed that SIP calls connected > > to the conference seemed to get out of sync. Basically, after perhaps 20 > > minutes or so in conference there was a 2 - 3 second delay between the > > time that one party spoke and the other party heard what was said. I have > > not tried ztdummy myself. Has anybody else seen this? > > According to > http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy > , ztrtc is supposted to fix the MeetMe delay. It compiles by default on > Linux 2.6 with newer versions of -HEAD. Only if you remove the #if 0 from around #define USE_RTC Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: meetme - conf-invalid
In article <[EMAIL PROTECTED]>, qrss <[EMAIL PROTECTED]> wrote: > Yes, meetme requires a clock source. You could try ztdummy. I tried > using an FXO card as a clock source and observed that SIP calls connected > to the conference seemed to get out of sync. Basically, after perhaps 20 > minutes or so in conference there was a 2 - 3 second delay between the > time that one party spoke and the other party heard what was said. I have > not tried ztdummy myself. Has anybody else seen this? Yes. Try the patch at http://bugs.digium.com/view.php?id=4252 to see whether it helps. Please post your results to that bug - thanks! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: meetme mute status
In article <[EMAIL PROTECTED]>, bdz <[EMAIL PROTECTED]> wrote: > hi, > > is there any way to figure out what the mute status > is of the meetme conference participants? > > i personally can no see any difference on the output: > kamikaze*CLI> meetme > Conf Num PartiesMarked Activity Creation > 5000 0002 N/A00:00:40 Static > * Total number of MeetMe users: 2 > kamikaze*CLI> > kamikaze*CLI> meetme list 5000 > User #: 1 Channel: SIP/fizik-c4eb > User #: 2 Channel: H323/ip$192.168.42.10:10659/14231 > kamikaze*CLI> > kamikaze*CLI> meetme mute 5000 1 > kamikaze*CLI> meetme list 5000 > User #: 1 Channel: SIP/fizik-c4eb > User #: 2 Channel: H323/ip$192.168.42.10:10659/14231 > kamikaze*CLI> > > i also can not see any mute/unmute event on the manager > interface only the join/leave events come. Neither of these features currently exists in MeetMe, although they are both on my ToDo list unless someone else beats me to it. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: Meetme echo cancellation
In article <[EMAIL PROTECTED]>, Steven Langley <[EMAIL PROTECTED]> wrote: > > I am using IAX2 softphones dialing into a meetme conference. In my > softphone I was forcing uses to click on a button when they wanted to > speak, enabling their microphone and disabling their speakers. This way > when a user was speaking they did not hear their voice half a second > later (because meetme mixes the voice and sends to everyone in the > conference). > > Now because of requirements there is a need for users not to have to > click a button when speaking (and have their microphones and speakers > enabled at all times) - much like Skype. How would I prevent a user > hearing their own voice half a second later? Using some kind of echo > cancellation? I am not sure that this is defined as echo though. > > Does anyone have any ideas? It's a hard problem, if your users are using speakers instead of headsets or normal phone handsets. On traditional speakerphones, it is the phone itself that has echo cancellation built-in, so that any speaker output that gets picked up by the microphone gets cancelled out in the signal that is sent along the phone line. So really, it is your soft phone that should be replicating the same functionality, and making sure it does not feed back audio from the speakers. I think It would be much more difficult to try to do it at the asterisk end. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme option 'b'
In article <[EMAIL PROTECTED]>, John Daragon <[EMAIL PROTECTED]> wrote:
> Hi;
>
> I've been looking for an arbitrary way of discovering when the last
> user has left a Meetme conference...
>
> It occurred to me that I could launch an agi script to keep watch over
> the conference and do something when the user count reaches zero... And
> of course, I can do that directly from the dialplan.
>
> But I was looking at app_meetme, and the docs say:
>
>
> * 'b' run AGI script specified in ${MEETME_AGI_BACKGROUND}
>
>
>o Default: conf-background.agi (Note: This does not work
> with non-Zap channels in the same conference)
>
>
> I can't see anything in the code to explain this; does anyone understand
> why it might be ?
To explain which part? That it doesn't work with non-Zap channels?
For Zap channels, the mixing is automatically done at the driver level
once MeetMe has told the driver which channels to mix.
For a non-Zap channel, a proxy Zap channel (pseudo) is created to
participate in the driver-level mix. The meetme thread on the channel
then enters a loop to copy audio back and forth between the non-Zap
channel and the proxy pseudo-channel.
When an AGI background script is specified, it runs INSTEAD OF the
copying loop mentioned above. Therefore there is nothing to move the
audio to and from the non-Zap channel.
Hope this helps!
Cheers
Tony
--
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Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: Meetme option Ax
In article <[EMAIL PROTECTED]>, tcchan <[EMAIL PROTECTED]> wrote: > Dear All, > > I am a bit confused with the Meetme option A and x. > > My intention is to close the conference room when the user calls > [newConf] hangup, I have the following line in extensions.conf: > > > [newConf] >exten => s,1,Answer >exten => s,2,MeetMe(,eAx) > > [enterConf] >exten => s,1,Answer >exten => s,2,MeetMe(,) > > UserA calls newConf and is assigned to room 100, userB calls enterConf > and entered room 100 when prompted. Then if userA hangup, the > conference room 100 is still active if userB still on hold. > > What did I miss? > > What should I do to make this work? I think you need to specify option x also on the second MeetMe (but not option A). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe Dialplan question
In article <[EMAIL PROTECTED]>, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: > Hi, > > is the following possible? I would like to transfer a call to my > "personal" MeetMe conference room and get transferred there > automatically as well. Currently I can transfer the call to the > conference, have to hangup and then call the conference number myself. I > would love to have this in one quick function. Look at the option 'G(context^exten^pri)' in the Dial application. > Moreover is there a way to disable the "You are currently the only > person in this conference" prompt for the first user? I know how to > enable/disable this for the following users but not for the first user. Specifying the 'q' flag to MeetMe disables it. However, it also disables all the enter/exit sounds and so on, so if you still want those you will have to either: - edit the code to provide another option to turn off that message, or - replace the sound file conf-onlyperson.gsm with a file of zero duration (not necessarily just an empty file - it might still need a header). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe Dialplan question
In article <[EMAIL PROTECTED]>, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: > > > Look at the option 'G(context^exten^pri)' in the Dial application. > > Thanks for the hint but I am not sure if this will help me. Either I am > too blind to see the solution or I stated the question in an unclear > way. :-) What I want is this: > > 1. Customer calls me or I call customer. > 2. In the middle of the call Aha, this was the bit I missed. > I decide to get an additional collegues in > the call. > > Usually I would put the first call on hold, call the collegue and then > press the conference button on my SNOM 360. Unfortunatly there seems to > be a problem with the SNOMs and Asterisk 1.2.x since audio on those > conferences get distorted after a few seconds. Therefore I need a > substitution for this using MeetMe. I thought about this: > > 3. I transfer the call to my "personal" MeetMe room. In this step I > would like not only the customer but also me to be connected to the > MeetMe room automatically. Basically I can continue to chat with the > customer without him noticing anything. > > 4. I now put the call on hold and call the collegue. If he wants to join > I simply transfer him to the room as well and can continue to do so with > other collegues. In order to return to the conference myself I now do > not need to call the conference number myself but simply return to the > call created in step 3. > > With the exception of step 3 everything seems easy. How can I solve this > with the G-option? I don't think you can. The G option was to put you both into the MeetMe room at the beginning of the call. I think the solution needs a little more thinking about Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] RE: Meetme admin option
After thinking about it for a few days, I realized that one way to prevent non-admin users from entering the conference room is to use an AGI script that actually performs the authentication. But, I would rather have the functionality built into the Meetme application. Are there is any plans in the near future for implementing this kind of control? Or should I consider posting a bounty for this? Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 -Original Message- From: Anish Basu [mailto:[EMAIL PROTECTED] Sent: Friday, October 21, 2005 4:16 PM To: asterisk-users@lists.digium.com Subject: Meetme admin option There is an Meetme command option 'a' for admin. I tried using this option and noticed that it allows users to login with the user pin as well as the admin pin. In my dialpan I have: exten => 700, 1, Meetme(500,Mas) And in meetme.conf, I have: conf => 500,1234, After dialing extension 700, I was able to login to the conference using the user pin '1234'. When I pressed the star key, I was presented with the voicemenu "Press 1 to mute/unmute yourself, 2 to lock/unlock this conference, or press 3 to eject the last user", which should only be for admin. Is there any way to restrict users from logging in unless they have the correct admin pin? Anish Basu Field Systems Engineer Softel, Inc. Phone: (732) 705-9202 Cell: (732) 312-6634 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meetme Conference-reg
In article <[EMAIL PROTECTED]>, Bartosz Piec <[EMAIL PROTECTED]> wrote: > I don't have app_meetme.so file neither in /usr/lib/asterisk/modules, > nor /usr/src/asterisk/apps. How to get it? You need to get, build and install zaptel on your system, and then rebuild Asterisk. Asterisk won't build app_meetme if it doesn't find zaptel on your system. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meetme Conference-reg
In article <[EMAIL PROTECTED]>, Bartosz Piec <[EMAIL PROTECTED]> wrote: > Tony Mountifield napisa³(a): > > You need to get, build and install zaptel on your system, and then > > rebuild Asterisk. > > ztdummy is enough? ztdummy is only a device driver. You also need the zaptel module. > Will building Asterisk break something in my working installation? :) Not if you do it properly and with understanding. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Meetme Conference-reg
In article <[EMAIL PROTECTED]>, Bartosz Piec <[EMAIL PROTECTED]> wrote: > Tony Mountifield napisa³(a): > > ztdummy is only a device driver. You also need the zaptel module. > > And this is this: > http://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz, right? I'm using > 1.0.9 version. Yes. > >>Will building Asterisk break something in my working installation? :) > > > > Not if you do it properly and with understanding. > > Backuping /etc/asterisk is enough? So long as you just do "make clean" and "make install", and DON'T do "make samples", then rebuilding won't disturb your /etc/asterisk directory; you just stop and restart asterisk and it will use the existing /etc/asterisk files. Of course, keeping a backup of /etc/asterisk is a good idea anyway... Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: MeetMe Video option
So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? If I get a chance this weekend I will take a look at the implementation and see what I can see. The mosaic thing should be pretty easy actually (really, just a scaling of each incoming stream and tiling them), but that won't work well for anything bigger than a 2x2 matrix, considering the bandwidth limitations of most users. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson Sent: Friday, January 30, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MeetMe Video option That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/"to do" list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes. I was hoping to look into someday but I'm swamped with 1000 other things of higher priority. I have been thinking though, of some ways it could be supported, starting with the simplest and easiest: 1. First, if only 2 of the phones in the conference are video phones, allow them to exchange their video with each other, while having all of the audio streams conferenced as usual. 2a. The next step could be having each videophone "rotate" which stream it was showing for a few seconds (20 seconds maybe?). i.e. you could have 3 video calls mixed with several audio-only calls. Initially video call #1 would show #2's image, #2 would show #3's image, #3 would show #1's image for a few seconds, then rotate them by 1. Of course you don't need to show your own! :) Actually, ours has a picture-in-picutre in the corner so you can see yourself all the time anyway. 2b. The other option instead of time-rotating the images would be to try to show the image of whoever was talking. That kind of sounds like a pain to me, but maybe it's doable. 3. The really fancy thing would be to have Asterisk decode all of the video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them and send them to each client. That REALLY sounds like a pain to me, but again, maybe it's doable. Right now I'd be pretty happy with 2a though. - Matt >Message: 3 >From: "Regovich, Timothy" <[EMAIL PROTECTED]> >To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> >Date: Fri, 30 Jan 2004 13:07:46 -0500 >Subject: [Asterisk-Users] MeetMe Video option >Reply-To: [EMAIL PROTECTED] > >Hello All: > >Has anyone configured a meetme conference to use video? >I have successfully used video phones to talk through *, but I cannot seem >to get video when those phones dial into a meetme conference. > >Is there something else that I need to be doing other than set the "v" flag >on my extension for the meetme app? > >Thanks, > >Tim > > ___ 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 -- Notice: This e-mail message, together with any attachments, contains information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ 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] Re: MeetMe Video option
Tim - I'm actually quite fond of the 2b solution in the video conference tools I've used (notably, Polycomm) where the video switches or camera pans depending on audio energy. This could work quite well with the existing features of "m" and "t". A combination of blending an audio-energy and 2x2 matrix would also be pretty slick, where maybe some callers would be "nailed-up" and never would leave the matrix, but the remaining panels would fluctuate based on last audio energy input. It would lead to interesting shouting matches in circumstances outside of "corporate" use of videophone technology. :-) It's a shame that * doesn't have Solaris as a well-supported (at all? anyone?) platform; there are the Sparc routines for fast video transforms built into the Sparc processor chipset that could do really cool and fast stuff for videoconferencing routines. So, I'm waiting for the iChat video client software to be supported via *; then I'll actually invest in the slick Apple firewire camera. JT At 4:02 PM -0500 1/30/04, Regovich, Timothy wrote: So you are actually getting the video to come out though? I am not getting any outbound video RTP traffic at all. What settings do you have? If I get a chance this weekend I will take a look at the implementation and see what I can see. The mosaic thing should be pretty easy actually (really, just a scaling of each incoming stream and tiling them), but that won't work well for anything bigger than a 2x2 matrix, considering the bandwidth limitations of most users. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson Sent: Friday, January 30, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: MeetMe Video option That's one of the things that's been on our (1control, I have nothing to do with Digium) wishlist/"to do" list that just hasn't gotten done yet. Currently, video in meetme is not supported. What we experience is the audio will conference with the other audio streams but the video just freezes. I was hoping to look into someday but I'm swamped with 1000 other things of higher priority. I have been thinking though, of some ways it could be supported, starting with the simplest and easiest: 1. First, if only 2 of the phones in the conference are video phones, allow them to exchange their video with each other, while having all of the audio streams conferenced as usual. 2a. The next step could be having each videophone "rotate" which stream it was showing for a few seconds (20 seconds maybe?). i.e. you could have 3 video calls mixed with several audio-only calls. Initially video call #1 would show #2's image, #2 would show #3's image, #3 would show #1's image for a few seconds, then rotate them by 1. Of course you don't need to show your own! :) Actually, ours has a picture-in-picutre in the corner so you can see yourself all the time anyway. 2b. The other option instead of time-rotating the images would be to try to show the image of whoever was talking. That kind of sounds like a pain to me, but maybe it's doable. 3. The really fancy thing would be to have Asterisk decode all of the video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them and send them to each client. That REALLY sounds like a pain to me, but again, maybe it's doable. Right now I'd be pretty happy with 2a though. - Matt Message: 3 From: "Regovich, Timothy" <[EMAIL PROTECTED]> To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> Date: Fri, 30 Jan 2004 13:07:46 -0500 Subject: [Asterisk-Users] MeetMe Video option Reply-To: [EMAIL PROTECTED] Hello All: Has anyone configured a meetme conference to use video? >I have successfully used video phones to talk through *, but I cannot seem >to get video when those phones dial into a meetme conference. > >Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, >Tim > ___ 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] Re: MeetMe Video option
Few pieces of info that might help us from re-inventing the wheel. What we are all taking about has a name in video conferencing circles. It is called an MCU, Multi Conferencing Unit. The OpenH323 project has an MCU called OpenMCU. Since the H323 support in * is based on OpenH323 perhaps it would be possible to tie in the OpenMCU into the MeetMe code. There is probably some lower level stuff int he h323 channel stuff as well that details what codecs are supported. My guess is that the * wrapper only advertises support for audio codecs which is part of why the video drops out when transfering to a MeetMe conference. The general mode of operation for an MCU is to key off of the audio signal. 2 x 2 grid is common with the video signals being sent from the currently active audio channels, or most recently active. Or a single larger video of the currently speaking source. This is the mode of OpenMCU. Some of the more sophisticated MCU servers allow individual users to select which views they get. I think this would be an extremely cool feature for * to support. I can think of several potential clients I could snag with such a setup. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting John Todd <[EMAIL PROTECTED]>: > Tim - >I'm actually quite fond of the 2b solution in the video conference > tools I've used (notably, Polycomm) where the video switches or > camera pans depending on audio energy. This could work quite well > with the existing features of "m" and "t". > >A combination of blending an audio-energy and 2x2 matrix would also > be pretty slick, where maybe some callers would be "nailed-up" and > never would leave the matrix, but the remaining panels would > fluctuate based on last audio energy input. It would lead to > interesting shouting matches in circumstances outside of "corporate" > use of videophone technology. :-) > >It's a shame that * doesn't have Solaris as a well-supported (at > all? anyone?) platform; there are the Sparc routines for fast video > transforms built into the Sparc processor chipset that could do > really cool and fast stuff for videoconferencing routines. > >So, I'm waiting for the iChat video client software to be supported > via *; then I'll actually invest in the slick Apple firewire camera. > > JT > > > At 4:02 PM -0500 1/30/04, Regovich, Timothy wrote: > > > >So you are actually getting the video to come out though? > >I am not getting any outbound video RTP traffic at all. What settings do > >you have? > > > >If I get a chance this weekend I will take a look at the implementation and > >see what I can see. > >The mosaic thing should be pretty easy actually (really, just a scaling of > >each incoming stream and tiling them), but that won't work well for > anything > >bigger than a 2x2 matrix, considering the bandwidth limitations of most > >users. > > > >Tim > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of Matt Lawson > >Sent: Friday, January 30, 2004 3:13 PM > >To: [EMAIL PROTECTED] > >Subject: [Asterisk-Users] Re: MeetMe Video option > > > > > >That's one of the things that's been on our (1control, I have nothing to > >do with Digium) wishlist/"to do" list that just hasn't gotten done yet. > > > >Currently, video in meetme is not supported. What we experience is the > >audio will conference with the other audio streams but the video just > >freezes. I was hoping to look into someday but I'm swamped with 1000 > >other things of higher priority. I have been thinking though, of some > >ways it could be supported, starting with the simplest and easiest: > > > >1. First, if only 2 of the phones in the conference are video phones, > >allow them to exchange their video with each other, while having all of > >the audio streams conferenced as usual. > > > >2a. The next step could be having each videophone "rotate" which stream > >it was showing for a few seconds (20 seconds maybe?). i.e. you could > >have 3 video calls mixed with several audio-only calls. Initially video > >call #1 would show #2's image, #2 would show #3's image, #3 would show > >#1's image for a few seconds, then rotate them by 1. Of course you > >don't need to show your own! :) Actually, ours has a > >picture-in-picutre in the corner so you can see yourself all the time > >anyway. > > > >2b. The other option instead of time-rotating the images would be to > &
[asterisk-users] Re: meetme ${DATETIME} variable update
On 1/19/07, nik600 <[EMAIL PROTECTED]> wrote:
Hi i am experiencing this problem:
MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE
exten => ,1,MeetMe(666|1Arxq)
exten => 9998,1,MeetMe(666|1Axq)
exten => 9997,1,MeetMe(666|1xq)
I make a conference between 3 person dialing
A dials
B dials 9998
C dials 9997
all works fine but the datetime won't be updated, it still remain for
example 13:40 until i do a complete restart of asterisk.
where can be the problem?
it seems that i've fixed using
Set(MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE)
before each meetme
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Re: [asterisk-users] Re: MeetMe and ChannelRedirect
On 17/05/07, Tony Mountifield <[EMAIL PROTECTED]> wrote: In article <[EMAIL PROTECTED]>, Rafael Vidal Aroca <[EMAIL PROTECTED]> wrote: > i'm trying to implement the following scenario: > > - A user calls number 700 > - Asterisk then dials to extensions 100, 200, 300, 400 and 500 > - And then bridges all calls to a conference room > > I tried to use MeetMe and ChannelRedirect, but seems that after > channel redirect nothing more is executed. So, this seem to work for the > caller and first called, but the others stay outside. > > Could anyone help or give me a hint? The way I did this kind of thing was like this: 1. Extension 700 calls an AGI script which generates a .call file in /var/spool/asterisk/outgoing for each of the calls to the other extensions. Extension 700 then drops into the Meetme room to wait for the others. 2. Each call file specifies a Local channel to make the call to the extension, and uses the Context, Extension and Priority fields to direct the answered call into the Meetme room. If any of the calls to the other extensions fails (e.g. busy), you don't get any notification of that. If you want such notification, you will need to get a lot more complex, probably involving a controlling process using the Manager API. Hope this helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] Re: MeetMe and ChannelRedirect
On 17/05/07, Andrew Furey <[EMAIL PROTECTED]> wrote: [nothing] Ugh, what happened there? must have clicked the wrong button. Sorry for the noise folks. Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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] RE: Meetme with video???
I'm interested in working on this project, please contact me if you plan or are actually working on this and we can probably coordinate something here. Regards, Shidan shidan at gmail On Fri, 17 Dec 2004 09:05:13 -0500, Noah Miller <[EMAIL PROTECTED]> wrote: > > I wonder if there is an application available, what would > > allow me to have a conference call (meetme) with video. > > Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this > functionality: > > http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+video+conferencing > > You can add to this bounty, if you want. I'm trying to convince the money > people at my company that we should add $500 to this. > > BTW: Is anybody working on this? > > ___ > 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] RE: Meetme with video???
Hi Noah, I have been contacted by 2 people but nothing so far. If you want to add $500 please email me your details and I'll add it to the wiki to co-ordinate this. I agree I'm really surprised why no one has shown more of an interest in video calls on asterisk yet. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, December 17, 2004 9:05 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Meetme with video??? > I wonder if there is an application available, what would > allow me to have a conference call (meetme) with video. Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this functionality: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing You can add to this bounty, if you want. I'm trying to convince the money people at my company that we should add $500 to this. BTW: Is anybody working on this? ___ 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] RE: Meetme with video???
[EMAIL PROTECTED] wrote: > Hi Noah, I have been contacted by 2 people but nothing so > far. If you want to add $500 please email me your details and > I'll add it to the wiki to co-ordinate this. > > I agree I'm really surprised why no one has shown more of an > interest in video calls on asterisk yet. It is certainly worthy of discussion, but perhaps not really that surprising; video has been the next big thing in telephony since the 50s. I think the price of entry has always scared folks off, not to mention that the kind of bandwidth and horsepower required by video makes audio seem a piece of cake. But I think the real barrier with video is the cultural environment. We have certain expectations when we're seeing video: years of television has trained us to expect a certain level of production. This means that video connections come with certain cultural expectations that audio does not. Please understand that I'm not saying video is bad, or useless, or unimportant, or whatever. I'm just observing the fact that the industry has been promising that video will be the standard in communication for over fifty years, and people still love their telephones. Consider: I could have a business conversation with you right now over the telephone, but as I'm still in my underwear, a video conversation would not be fun for you at all! The interest is there, but perhaps everyone's got enough to do with audio to get into worrying about video just yet. It's not that no one wants it, it's more a matter of priorities. Again, I'm not trying to put a value judgement on it, merely to speculate on why the interest is not as high as one might expect. Regards, Jim. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Noah Miller > Sent: Friday, December 17, 2004 9:05 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] RE: Meetme with video??? > >> I wonder if there is an application available, what would >> allow me to have a conference call (meetme) with video. > > Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this > functionality: > > http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid > eo+conferencing > > You can add to this bounty, if you want. I'm trying to convince the > money people at my company that we should add $500 to this. > > BTW: Is anybody working on this? > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ 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] RE: Meetme with video???
Hi Tom, thanks for the heads up, it looks interesting but I'm not sure whether it would be easier for someone to start from scratch rather than modify reflector. I have found these 2 links http://www.geektimes.com/michael/CU-SeeMe/faqs/reflectors.html https://sourceforge.net/projects/cuseeme/ Cheers, Dean -Original Message- From: Tom Chandler [mailto:[EMAIL PROTECTED] Sent: Saturday, December 18, 2004 12:03 PM To: dean collins Subject: Fw: [Asterisk-Users] RE: Meetme with video??? Dean, I read this with great interest, much as you have. My question is there has been software GPL on the net for several years that support this type of application. I used with cu-see-me. It is called a reflector, written in "C", and allows both voice and video as in a meetme conference. I am NOT strong in C, but someone who is could take some of this existing code and put it into Asterisk. A lot of the work has already been done, now to include it in Asterisk. If you need more information, please let me know. I think I have the source on this software somewhere here. Development stopped about two years ago. Tom Chandler - Original Message - From: "dean collins" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, December 18, 2004 10:37 AM Subject: RE: [Asterisk-Users] RE: Meetme with video??? > Hi Noah, I have been contacted by 2 people but nothing so far. If you > want to add $500 please email me your details and I'll add it to the > wiki to co-ordinate this. > > I agree I'm really surprised why no one has shown more of an interest in > video calls on asterisk yet. > > > Cheers, > Dean > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Noah > Miller > Sent: Friday, December 17, 2004 9:05 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] RE: Meetme with video??? > > > I wonder if there is an application available, what would > > allow me to have a conference call (meetme) with video. > > Nope, AFAIK there's nothing yet. There is a bounty of $2000 for this > functionality: > > http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid > eo+conferencing > > You can add to this bounty, if you want. I'm trying to convince the > money people at my company that we should add $500 to this. > > BTW: Is anybody working on this? > > ___ > 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 > > > --- > Scanned by Bayou Internet for all known viruses. > http://www.bayou.com > > ___ 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] RE: Meetme with video???
Hi, > -Original Message- > Hi Noah, I have been contacted by 2 people but nothing so far. If you > want to add $500 please email me your details and I'll add it to the > wiki to co-ordinate this. > > I agree I'm really surprised why no one has shown more of an > interest in > video calls on asterisk yet. I have been looking at video with asterisk but it is unfortunately not high enough on my priorities list to really add to this either from a cash or resource point of view. I do hope this will evolve in some way though... Florian ___ 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] RE: Meetme with video???
Florian Overkamp wrote: Hi, -Original Message- Hi Noah, I have been contacted by 2 people but nothing so far. If you want to add $500 please email me your details and I'll add it to the wiki to co-ordinate this. I agree I'm really surprised why no one has shown more of an interest in video calls on asterisk yet. I have been looking at video with asterisk but it is unfortunately not high enough on my priorities list to really add to this either from a cash or resource point of view. I do hope this will evolve in some way though... Florian I am not in the position to finance it yet, but I can tell you what I found out about the business model. In Taiwan English learning is the number one of income Every school teaches English with more or less good teachers. Even adults are now trying to catch up with English, so that they do not appear as stupid beside their kids. ADSL is in each household with at leas 2M/256k, VoIP is also used. If we can make classrooms with phones, for people who just want to talk and we could add video for the ones who could afford it. Classrooms could be used more or less 18 hours per day Video is one thing what makes the class more attractive. The second is a whiteboard and push urls, Again, I would love to be in the position to pay for such application based on Asterisk, but unfortunately I am not (yet) However, if I hear more details from people who are able to make it possible, I might be able to get the foundings for it. bye Ronald ___ 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] Re: Meetme option 'b'
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>, John Daragon <[EMAIL PROTECTED]> wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' — run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
To explain which part? That it doesn't work with non-Zap channels?
For Zap channels, the mixing is automatically done at the driver level
once MeetMe has told the driver which channels to mix.
For a non-Zap channel, a proxy Zap channel (pseudo) is created to
participate in the driver-level mix. The meetme thread on the channel
then enters a loop to copy audio back and forth between the non-Zap
channel and the proxy pseudo-channel.
When an AGI background script is specified, it runs INSTEAD OF the
copying loop mentioned above. Therefore there is nothing to move the
audio to and from the non-Zap channel.
Hope this helps!
It does, indeed ! Thanks for the succinct explanation.
I owe you a beer.
jd
--
John Daragon [EMAIL PROTECTED]
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Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK
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RE: [Asterisk-Users] Re: MeetMe Dialplan question
Hi Tony, > Look at the option 'G(context^exten^pri)' in the Dial application. Thanks for the hint but I am not sure if this will help me. Either I am too blind to see the solution or I stated the question in an unclear way. :-) What I want is this: 1. Customer calls me or I call customer. 2. In the middle of the call I decide to get an additional collegues in the call. Usually I would put the first call on hold, call the collegue and then press the conference button on my SNOM 360. Unfortunatly there seems to be a problem with the SNOMs and Asterisk 1.2.x since audio on those conferences get distorted after a few seconds. Therefore I need a substitution for this using MeetMe. I thought about this: 3. I transfer the call to my "personal" MeetMe room. In this step I would like not only the customer but also me to be connected to the MeetMe room automatically. Basically I can continue to chat with the customer without him noticing anything. 4. I now put the call on hold and call the collegue. If he wants to join I simply transfer him to the room as well and can continue to do so with other collegues. In order to return to the conference myself I now do not need to call the conference number myself but simply return to the call created in step 3. With the exception of step 3 everything seems easy. How can I solve this with the G-option? > Specifying the 'q' flag to MeetMe disables it. However, it also > disables all the enter/exit sounds and so on, so if you still want > those you will have to either: Yep found that but as you said it disables all sounds. > - edit the code to provide another option to turn off that message, or Can't be too hard. I will have a look at the code an provide a patch. > - replace the sound file conf-onlyperson.gsm with a file of zero > duration (not necessarily just an empty file - it might still need > a header). Since I want this for usual conferences this is not an option I guess. :-) Kind regards, JP ___ --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] Re: MeetMe Dialplan question
On Saturday, January 21, 2006 1:44 PM Tony Mountifield wrote: > - edit the code to provide another option to turn off that message, or http://bugs.digium.com/view.php?id=6316 Regards, JP ___ --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] Re: MeetMe Dialplan question
> I think the solution needs a little more thinking about I am reliefed. I almost thought I had missed something that obvious... ___ --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] Re: Meetme Conference-reg
Tony Mountifield napisał(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? Will building Asterisk break something in my working installation? :) -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Meetme Conference-reg
On Mon, 2005-11-07 at 10:07 +0100, Bartosz Piec wrote: > Tony Mountifield napisał(a): > > You need to get, build and install zaptel on your system, and then > > rebuild Asterisk. > > ztdummy is enough? > it is for app_meetme, you just need a timing source that is external. app_conference doesnt require this however, but there are some limitations, such as no dtmf while in the conference (exiting, admin menu, etc). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Meetme Conference-reg
No - just make sure you DO NOT type make samples. regards, Jenn - Original Message - From: "Bartosz Piec" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Monday, November 07, 2005 8:07 PM Subject: Re: [Asterisk-Users] Re: Meetme Conference-reg > Tony Mountifield napisał(a): > > You need to get, build and install zaptel on your system, and then > > rebuild Asterisk. > > ztdummy is enough? > > Will building Asterisk break something in my working installation? :) > > -- > Best regards, > Bartosz Piec > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Meetme Conference-reg
Tony Mountifield napisał(a): ztdummy is only a device driver. You also need the zaptel module. And this is this: http://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz, right? I'm using 1.0.9 version. Will building Asterisk break something in my working installation? :) Not if you do it properly and with understanding. Backuping /etc/asterisk is enough? -- Best regards, Bartosz Piec ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe - Causes * to crash :/
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > Anyone ever seen MeetMe cause * to crash? Specifically, it happens > consistantly if someone begins to enter a conference and then decides to > hangup while Allison is introducing them - like playing back > "conf-onlyperson". This has been seen with the MeetMe participant connecting > via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't > seen it). > > The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards, > Mandriva 2006 Free. I can confirm that asterisk 1.2.1 with zaptel 1.2.1 doesn't freeze on SIP channel. -- Tomislav Parcina tparcina#lama.hr ___ --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] Re: * Meetme Freeze patch found
Forgoe the patch, just upgrade to 1.2.6. The changelog lists it as a fix from 1.2.5 to 1.2.6. >I'm a bit newbie, could you tell me how to i apply the patch? > >Thanks in advance >Marco Mouta > >On 3/27/06, Benoit Panizzon <[EMAIL PROTECTED]> wrote: >> On Friday 24 March 2006 16:05, Benoit Panizzon wrote: >> > Hi all >> > >> > Apparently there is a patch for those 1.2.4/5 MeetMe Freezes: >> > >> > http://bugs.digium.com/view.php?id=5884 >> > >> > Haven't tried it out yet. >> >> I can now confirm: No freezes/crashes anymore since I applied the patch. >> >> -Benoit- Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --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] Re: meetme conference latency degrades...
In article <[EMAIL PROTECTED]>, Chris Stenton <[EMAIL PROTECTED]> wrote: > This is a known problem and it does not matter what zaptel timer you use. A > solution is available in 'svn head' by using > > asterisk.conf > internal_timing = yes > > OR > Enable internal timing support (-I) > > on the command line. I don't know if this has been backported to the stable > branch. It hasn't, specifically, but the required changes are not large, and it is easy to apply the changes by hand; I do. Go to http://bugs.digium.com/view.php?id=5374 and download the last asynchronous patch, 2005-10-04-3-asynchronous.patch Then apply it by hand to channel.c, and also to app_milliwatt.c and app_sms.c if you happen to be using those applications. I don't think the app_chanspy.c patch is required any more. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] RE: meetme conference latency degrades...
I think you need to upgrade to the latest Asterisk. Your version is pretty ancient. > We are using v1.0.8 - Original Message - From: "Michael George" <[EMAIL PROTECTED]> To: Sent: Thursday, May 04, 2006 2:48 AM Subject: [Asterisk-Users] meetme conference latency degrades... > We have recently started making more frequent use of the meetme > conference of our * system. > > We are using v1.0.8 with a 2.6.11 kernel on our system. > > We generally have 4 callers in it: two with the gsm codec and 2 with g729. > Initially, the conference works fine and there is little latency. After > about 15min., though, the latency is very noticable and by 25min it's > unbearable. > > If we all leave the conference and return, the latency is unnoticable > again. > > The load on the box is minimal, and only our meetme is running most of > the time. Checking system load with top shows 0.1 or less. > > We have no digium hardware and use ztdummy for our timing device. > zttest yields results generally in the area of 99.96%, but about 3-4% > will be as low as 95%. > > In much smaller systems with Digium hardware, the accuracy is never > below 99.98% and is often 100%. > > Is this apparent inaccuracy of the ztdummy timer likely the cause of the > increasing latency in our meetme conference? > > Is there any way to improve it? > > Thank you, in advance, for any help. > > -- > -M ___ --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] Re: MeetMe announcements and SIP channels
In article <[EMAIL PROTECTED]>, Mike <[EMAIL PROTECTED]> wrote: > Just curious if anyone knows of any hacks to enable announce entry/exit > in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i > option will not work with SIP. The |i option does indeed work with SIP. You do have to have the zaptel driver loaded, and either ztdummy or a card driver, AND asterisk must load chan_zap.so Meetme won't work at all without zaptel, but if you want entry/exit announcements and/or recording to work, you also need chan_zap.so If you have done the above and it still doesn't work, what behaviour and log messages do you observe? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe Listen Only flag (|m)
(top posting for brevity, but original post included below, as it was over seven weeks ago) I've at last updated the patches for both trunk and 1.2, and posted them to Mantis at http://bugs.digium.com/view.php?id=6731 Cheers Tony In article <[EMAIL PROTECTED]>, Tony Mountifield <[EMAIL PROTECTED]> wrote: > In article <[EMAIL PROTECTED]>, > Tony Mountifield <[EMAIL PROTECTED]> wrote: > > In article <[EMAIL PROTECTED]>, > > Dan Austin <[EMAIL PROTECTED]> wrote: > > > Tony wrote: > > > > I should tidy it up and submit it, but haven't got round to it :-( > > > > > > Let us know if you can. I'm already maintaining a grocery list > > > of patches to make MeetMe viable in my orginization, so one more > > > won't kill me. > > > > I should be able do so this weekend. That's the plan, anyway :-) > > > > I'll post the Mantis bug# when I've submitted it. > > OK, just to reassure people I didn't forget, I've now produced a patch > for trunk and another for the 1.2 branch, by porting my changes across > from the version I had, which was based on 1.0. > > However, I think I ought to check they compile and run before I submit > them! I've run out of time to do that which weekend, so it will be a > couple of days. > > If anyone else would like to try them out any quicker, please email me > and I'll send you copies. > > Just to summarise what these patches provide: > > 1. The muting logic in the conference loop is tidied up, so that muting > and unmuting is done according to the flag states near the top of the > loop, and the DTMF muting/unmuting codes just set or clear the flags. > > 2. The 'm' flag now means "initially muted", but allows the user to be > unmuted from the command line. Users cannot unmute themselves if they > were muted from the command line, only if they muted themselves. > > 3. The new 'l' flag means "listen only" and is what the 'm' flag used > to be - unmuting is not possible. > > 4. Manager API events are generated when a user is muted or unmuted by > admin or themselves. > > 5. The code '*' in the admin or user menus generates an API event which > can be used by a user to attract the attention of an operator (e.g. a > muted user who wishes to speak). > > Cheers > Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe problem - some parameters ignored
In article <[EMAIL PROTECTED]>, Jim Archer <[EMAIL PROTECTED]> wrote: > Hi All... > > I set up a conference bridge using MeetMe. It works nicely, except that it > seems that certain parameters I give it are ignored or else don't work. > > Here is the line from my dial plan: > > exten => 6500,1,absolutetimeout,0 > exten => 6500,2,MeetMe,100|ciMpPs|1234 > > The MOH and * work, but users are not announced when they join or leave > and the pin is not requested. Maybe I am misunderstanding what these > are supposed to do? You need to read about the difference between CVS-HEAD (development version) and CVS-STABLE (the 1.0.x series). Some of the above options (including 'i'), and also the 'r' option in your other posting, only exist in the development version, not in the 1.0.x versions, as they were added after the 1.0 feature freeze was made. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: MeetMe problem - some parameters ignored
In article <[EMAIL PROTECTED]>, Tony Mountifield <[EMAIL PROTECTED]> wrote: > > and also the 'r' option in your other posting, Oops, my mistake. The posting about the 'r' option was from Jason Walker, not Jim Archer. But the same answer still applies. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: MeetMe Listen Only flag (|m)
In article <[EMAIL PROTECTED]>, Dan Austin <[EMAIL PROTECTED]> wrote: > One of the features that I thought would be popular with the Web-MeetMe > suite is the ability to start all non-admin callers in a muted state and > selectively unmute them. For example any large conference that is > of an announcment nature with a Q&A session. > > It's probably a feature I should have tested better, but I just > discovered > that a caller that is joined to a MeetMe with the |m flag cannot be > unmuted with command *cli>meetme unmute $confno $caller-no > > Is anyone using MeetMe for such conferences, and if so what combination > of flags do you use, and how do you change their state from Listen Only > to Talk&Listen? I needed the same functionality. There wasn't a way to do it in the current version of MeetMe. Also, the current muting logic is a bit of a mess. I reworked the muting logic, and changed MeetMe so that an 'l' flag meant listen-only (like the current 'm'), and an 'm' flag meant initially-muted. I also put in Manager Events to inform when a user was muted or unmuted. I should tidy it up and submit it, but haven't got round to it :-( Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe Listen Only flag (|m)
In article <[EMAIL PROTECTED]>, Dan Austin <[EMAIL PROTECTED]> wrote: > Tony wrote: > > I should tidy it up and submit it, but haven't got round to it :-( > > Let us know if you can. I'm already maintaining a grocery list > of patches to make MeetMe viable in my orginization, so one more > won't kill me. I should be able do so this weekend. That's the plan, anyway :-) I'll post the Mantis bug# when I've submitted it. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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[2]: [Asterisk-Users] Re: MeetMe Dialplan question
Hello,
Saturday, January 21, 2006, 7:34:50 PM, you wrote:
> 3. I transfer the call to my "personal" MeetMe room. In this step I
> would like not only the customer but also me to be connected to the
> MeetMe room automatically. Basically I can continue to chat with the
> customer without him noticing anything.
> 4. I now put the call on hold and call the collegue. If he wants to join
> I simply transfer him to the room as well and can continue to do so with
> other collegues. In order to return to the conference myself I now do
> not need to call the conference number myself but simply return to the
> call created in step 3.
> With the exception of step 3 everything seems easy. How can I solve this
> with the G-option?
What is the problem with step 3?
See this example as basis for modifications:
http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro
Use blind transfer and macro as above and you'll get this.
With step 4 we have tricky thing:
exten => _#901!,1,Dial(${EXTEN:4}, 10, G(meet^s-ENTER^1)
[meet]
exten => s-ENTER,1,Goto(s-HOLDER,1) <-- Here we get one channel
exten => s-ENTER,2,Goto(s-COLLEGUE) <-- And here we get other channel
exten => s-HOLDER,1,Hangup() <-- Leave this call, your collegue is here
exten => s-COLLEGUE,1,MeetMe(${CONFHOLDER},dwx) <-- Retrieve CONFHOLDER from
somewhere
exten => s-COLLEGUE,2,Hangup()
There may be minor mistakes, but I tried such dialplan and it works.
Hope this helps you.
--
Best regards,
Alexander Chemeris
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[Asterisk-Users] Re: MeetMe Listen Only flag (|m)
In article <[EMAIL PROTECTED]>, Tony Mountifield <[EMAIL PROTECTED]> wrote: > In article <[EMAIL PROTECTED]>, > Dan Austin <[EMAIL PROTECTED]> wrote: > > Tony wrote: > > > I should tidy it up and submit it, but haven't got round to it :-( > > > > Let us know if you can. I'm already maintaining a grocery list > > of patches to make MeetMe viable in my orginization, so one more > > won't kill me. > > I should be able do so this weekend. That's the plan, anyway :-) > > I'll post the Mantis bug# when I've submitted it. OK, just to reassure people I didn't forget, I've now produced a patch for trunk and another for the 1.2 branch, by porting my changes across from the version I had, which was based on 1.0. However, I think I ought to check they compile and run before I submit them! I've run out of time to do that which weekend, so it will be a couple of days. If anyone else would like to try them out any quicker, please email me and I'll send you copies. Just to summarise what these patches provide: 1. The muting logic in the conference loop is tidied up, so that muting and unmuting is done according to the flag states near the top of the loop, and the DTMF muting/unmuting codes just set or clear the flags. 2. The 'm' flag now means "initially muted", but allows the user to be unmuted from the command line. Users cannot unmute themselves if they were muted from the command line, only if they muted themselves. 3. The new 'l' flag means "listen only" and is what the 'm' flag used to be - unmuting is not possible. 4. Manager API events are generated when a user is muted or unmuted by admin or themselves. 5. The code '*' in the admin or user menus generates an API event which can be used by a user to attract the attention of an operator (e.g. a muted user who wishes to speak). Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] RE: MeetMe freezes machine with Junghanns
Dollars to donuts it is related to these two posts, but no one seems to know
where or why it happens - this issue doesn't seem to be related to one
specific piece of hardware:
Post 1)
*
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
"conf-onlyperson". This has been seen with the MeetMe participant connecting
via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't
seen it).
The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards,
Mandriva 2006 Free.
Symptoms of the crash: once the participant hangs up, the CLI seems to
freeze. One more call instance can be initiated, and the system will seize
within seconds (for instance, an audio prompt will deteriorate and then stop
dead). This behavior reminds me of the memory leak issue and time bomb bug,
perhaps they do the same damage as this.
Solution right now is to disable MeetMe, which isn't a solution as much as
an amputation. Anyways, here is the CLI output, note the WARNING:
alpha*CLI>
-- Executing Goto("SIP/Brent_ring-4473", "conferences|900|1") in new stack
-- Goto (conferences,900,1)
-- Executing MeetMe("SIP/Brent_ring-4473", "900|sMi|1234") in new stack ==
Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '900'
-- Recording
-- Playing 'vm-rec-name' (language 'en') Mar 15 16:44:38 WARNING[24014]:
file.cL584 ast_readaudio_callback: Failed to write frame
-- Playing 'conf-onlyperson' (language 'en') Alpha*CLI>
*
Post 2)
*
Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if
I hang up during Playing 'conf-onlyperson' my machine freezes. It's not a
GSM Enconding problem as I suspected first, this happens with every
encoding.
magma*CLI>
-- Executing Answer("SIP/11-9d7c", "") in new stack
-- Executing MeetMe("SIP/11-9d7c", "555") in new stack
-- Created MeetMe conference 1023 for conference '555'
-- Playing 'conf-onlyperson' (language 'de') magma*CLI>
Freeze!
Any other who can reproduce that freeze?
Kernel 2.6.15 / * 1.2.5 / ztdummy 1.2.4
**
Sincerely,
Brent A. Torrenga
[EMAIL PROTECTED]
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com
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[Asterisk-Users] Re: MeetMe Call Out to invite
In article <[EMAIL PROTECTED]>, welemon lee <[EMAIL PROTECTED]> wrote: > hi all, > > is there a kind of application can let asterisk call out > fellows, and invite them to come to join the meetme. > > these fellows do not need to call in asterisk , just wait for a call. You could try adapting the patch from http://bugs.digium.com/view.php?id=3405 It's quite old, so you will most likely have to apply it by hand. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
In article <[EMAIL PROTECTED]>, Dave Miller <[EMAIL PROTECTED]> wrote: > We upgraded our asterisk server to 1.2.18 last night to pick up the > security update. Since then, any calls coming in on IAX2 links get > dropped if they try to enter a MeetMe conference room. > > The log shows this: > > Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should > never be called! Hanging up. > > I've temporarily worked around it by switching our inbound provider to > use SIP instead of IAX, but that's not an ideal solution. What was the last version that successfully worked for you? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
In article <[EMAIL PROTECTED]>, Dinesh Nair <[EMAIL PROTECTED]> wrote: > > is there a patch for this against 1.2.18 ? it would sure help those who're > tracking the release tarballs instead of having to svn and compile it. Have a look at: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_iax2.c?r1=61866&r2=62037 The change is so trivial is would be easy to apply by hand. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: MeetMe - new e and E flags?
In article <[EMAIL PROTECTED]>, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > If it's a pin-required conference, you will hear the conference number > prior to being prompted to enter the associated pin. Obviously, in > this case, any such conference would be static, so the pin would be > pre-assigned in the config file. This might be useful if you ran a > number of conferences, but did not want just anybody to be able to > access them (i.e. in order to access the conferences, possibly > dial-able from anywhere, you had to know the associated pin). > > You can also select an empty dynamic conference, with pin, by > combining the flags 'eD', in which case you will be told the > conference number prior to you specifying the pin. Or you could > simply select an empty dynamic conference (no pin), with flags 'ed'. I'm trying hard to understand the usefulness of these features. It looks like, from what I've read here, if you dial an extension that routes to MeetMe(e), it will put you in an empty conference and tell you the number. Presumably for anyone else to join the same conference, you then have to tell them the number, e.g. by email, IM or another phone call, and they then have to dial a different extension which routes to MeetMe(without e). And if the empty conference also has a PIN, does the first user need a list of conference numbers to PINs so he can enter the correct PIN when told the conference number? This all seems rather cumbersome, and I haven't had the chance to experiment with this feature yet, so the above probably highlights both (a) my lack of understanding, and (b) the lack of documentation! Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: MeetMe - new e and E flags?
In article <[EMAIL PROTECTED]>, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Thursday 15 April 2004 03:01, Tony Mountifield wrote: > > > This all seems rather cumbersome, and I haven't had the chance to > > experiment with this feature yet, so the above probably highlights > > both (a) my lack of understanding, and (b) the lack of > > documentation! > > If the feature doesn't make any sense to you, then don't use it. I'd much rather be educated so that it does make sense to me. Then I too might find it useful. > For a customer of ours, though, it was necessary to have this feature. Could you explain why? I'm sure that knowing why it was important to them would help a lot in understanding (a) its purpose, and (b) the correct way to use it. I didn't know until you said so that it was you who implemented this feature - please could I ask you to add something to the wiki about it? > I would suggest actually trying out the feature a couple times, if > your goal is to learn how to use it. That's what I've been doing this evening, with not much success. I have the following in extensions.conf: exten => 4003,1,Answer exten => 4003,2,Wait(1) exten => 4003,3,MeetMe(|eMp) exten => 4003,4,Hangup exten => 4004,1,Answer exten => 4004,2,Wait(1) exten => 4004,3,MeetMe(|EMp) exten => 4004,4,Hangup And the following in meetme.conf: conf => conf => 1234 conf => 2345,9938 conf => 3131 conf => 4242 conf => 5353 conf => 6464 conf => 7575 conf => 8686 conf => 9797 I observed exactly the same behaviour whether I dialled 4003 or 4004: The first phone to dial was told I was in conference , and then received Music On Hold. The second and subsequent phones to dial 4003 or 4004 just received silence, not a new empty conference. No announcement, no MoH. "sip show channels" showed a new channel that did not disappear on hangup. I suspect this is a bug of some sort, but it's possible I have just done something wrong. Cheers, Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ 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] Re: MeetMe 'i' option not working correctly?
In article <[EMAIL PROTECTED]>,
Jon Webster <[EMAIL PROTECTED]> wrote:
> I'm running 2.4.5 and app_meetme never plays conf-hasleft or
> conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
> determine the cause. Any suggestions are greatly appreciated.
>
> exten => 600,1,MeetMe(600|i) I get the following:
>
> -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack
> == Parsing '/etc/asterisk/meetme.conf': Found
> Mar 8 06:13:53 WARNING[5197]: channel.c:2535 ast_request: No channel
> type registered for 'zap'
> Mar 8 06:13:53 WARNING[5197]: app_meetme.c:461 build_conf: Unable to
> open pseudo channel - trying device
The above messages indicate that chan_zap.so isn't loaded. Possibly it
isn't even built. You need to build *and install* zaptel before starting
to build Asterisk. Asterisk will find the zaptel libraries and will
build chan_zap.
Or maybe you just did what someone else did the other day: added a noload
statement for chan_zap in modules.conf because you didn't have any zaptel
hardware. You need chan_zap for MeetMe, and if you haven't got any zaptel
hardware, you also need to make sure ztdummy is loaded and running correctly.
Hope this helps.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Re: MeetMe Listen Only flag (|m)
On Wed, 2006-01-18 at 11:05 +, Tony Mountifield wrote: [snip] > I reworked the muting logic, and changed MeetMe so that an 'l' flag meant > listen-only (like the current 'm'), and an 'm' flag meant initially-muted. > > I also put in Manager Events to inform when a user was muted or unmuted. > > I should tidy it up and submit it, but haven't got round to it :-( Your patch is a nice enhancement so if you could find the time to submit it that would be great (hopefully compatible with the 1.2 branch too :) 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] Re: MeetMe Listen Only flag (|m)
Tony wrote: > I needed the same functionality. There wasn't a way to do it in the > current version of MeetMe. Also, the current muting logic is a bit > of a mess. I concur. > I reworked the muting logic, and changed MeetMe so that an 'l' flag > meant listen-only (like the current 'm'), and an 'm' flag meant > initially-muted. Now that's an idea I think I could impliment... > I also put in Manager Events to inform when a user was muted or > unmuted. > I should tidy it up and submit it, but haven't got round to it :-( Let us know if you can. I'm already maintaining a grocery list of patches to make MeetMe viable in my orginization, so one more won't kill me. Dan ___ --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: Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote: > What is the problem with step 3? > > See this example as basis for modifications: > http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro Unless I have terribly misunderstood that macro, that is basically the same thing I am doing now, is it not? Simply transfer the customer to a conference room (I might have a look into the automatically determined conf room number), then transfer all collegues in there as well and finally jump in myself. It is however not quite what I described in step 3. Thanks, JP ___ --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: Re[2]: [Asterisk-Users] Re: MeetMe Dialplan question
On 1/23/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote: > On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote: > > What is the problem with step 3? > > > > See this example as basis for modifications: > > http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro > > Unless I have terribly misunderstood that macro, that is basically the > same thing I am doing now, is it not? Simply transfer the customer to a > conference room (I might have a look into the automatically determined > conf room number), then transfer all collegues in there as well and > finally jump in myself. It is however not quite what I described in step 3. Yes, that's so. I tested this macro with SIP-softphones and it works. May be this is the simplest way to do what you want. And this is a good start point for modifications. -- Good luck, Alexander Chemeris ___ --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] RE: MeetMe freezes machine with Junghanns
On 3/23/06, Brent Torrenga <[EMAIL PROTECTED]> wrote:
> Dollars to donuts it is related to these two posts, but no one seems to know
> where or why it happens - this issue doesn't seem to be related to one
> specific piece of hardware:
>
> Post 1)
>
> *
> Anyone ever seen MeetMe cause * to crash? Specifically, it happens
> consistantly if someone begins to enter a conference and then decides to
> hangup while Allison is introducing them - like playing back
> "conf-onlyperson". This has been seen with the MeetMe participant connecting
> via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't
> seen it).
>
> The box is * 1.2.5, Zaptel 1.2.4, a TDM400P loaded with 3xFXO cards,
> Mandriva 2006 Free.
>
> Symptoms of the crash: once the participant hangs up, the CLI seems to
> freeze. One more call instance can be initiated, and the system will seize
> within seconds (for instance, an audio prompt will deteriorate and then stop
> dead). This behavior reminds me of the memory leak issue and time bomb bug,
> perhaps they do the same damage as this.
>
> Solution right now is to disable MeetMe, which isn't a solution as much as
> an amputation. Anyways, here is the CLI output, note the WARNING:
>
> alpha*CLI>
> -- Executing Goto("SIP/Brent_ring-4473", "conferences|900|1") in new stack
> -- Goto (conferences,900,1)
> -- Executing MeetMe("SIP/Brent_ring-4473", "900|sMi|1234") in new stack ==
> Parsing '/etc/asterisk/meetme.conf': Found
> -- Created MeetMe conference 1023 for conference '900'
> -- Recording
> -- Playing 'vm-rec-name' (language 'en') Mar 15 16:44:38 WARNING[24014]:
> file.cL584 ast_readaudio_callback: Failed to write frame
> -- Playing 'conf-onlyperson' (language 'en') Alpha*CLI>
> *
>
> Post 2)
>
> *
> Thank you for the hint. Now finaly I can 100% reproduce the problem. Yes, if
> I hang up during Playing 'conf-onlyperson' my machine freezes. It's not a
> GSM Enconding problem as I suspected first, this happens with every
> encoding.
>
> magma*CLI>
>-- Executing Answer("SIP/11-9d7c", "") in new stack
>-- Executing MeetMe("SIP/11-9d7c", "555") in new stack
>-- Created MeetMe conference 1023 for conference '555'
>-- Playing 'conf-onlyperson' (language 'de') magma*CLI>
>
> Freeze!
>
There's been two very recent commits (one less than an hour ago) that
may very well correct your issues.
--
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http://www.btwtech.com/
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Re: [Asterisk-Users] RE: MeetMe freezes machine with Junghanns
On Thursday 23 March 2006 22:14, BJ Weschke wrote: > There's been two very recent commits (one less than an hour ago) that > may very well correct your issues. The patch at http://bugs.digium.com/view.php?id=5884 fixes the problem! Cheers, Henning Holtschneider -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 pgpoDln7bEb6t.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped
Tony Mountifield wrote on 4/26/07 12:26 PM: > In article <[EMAIL PROTECTED]>, > Dave Miller <[EMAIL PROTECTED]> wrote: >> We upgraded our asterisk server to 1.2.18 last night to pick up the >> security update. Since then, any calls coming in on IAX2 links get >> dropped if they try to enter a MeetMe conference room. >> >> The log shows this: >> >> Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should >> never be called! Hanging up. >> >> I've temporarily worked around it by switching our inbound provider to >> use SIP instead of IAX, but that's not an ideal solution. > > What was the last version that successfully worked for you? 1.2.17. But the problem has been found and fixed (see my other post) -- Dave Miller http://www.justdave.net/ System Administrator, Mozilla Corporation http://www.mozilla.com/ Project Leader, Bugzilla Bug Tracking System http://www.bugzilla.org/ ___ --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] Re: MeetMe - new e and E flags?
On Thursday 15 April 2004 03:01, Tony Mountifield wrote: > In article <[EMAIL PROTECTED]>, > > Tilghman Lesher <[EMAIL PROTECTED]> wrote: > > If it's a pin-required conference, you will hear the conference > > number prior to being prompted to enter the associated pin. > > Obviously, in this case, any such conference would be static, so > > the pin would be pre-assigned in the config file. This might be > > useful if you ran a number of conferences, but did not want just > > anybody to be able to access them (i.e. in order to access the > > conferences, possibly dial-able from anywhere, you had to know > > the associated pin). > > > > You can also select an empty dynamic conference, with pin, by > > combining the flags 'eD', in which case you will be told the > > conference number prior to you specifying the pin. Or you could > > simply select an empty dynamic conference (no pin), with flags > > 'ed'. > > I'm trying hard to understand the usefulness of these features. It > looks like, from what I've read here, if you dial an extension that > routes to MeetMe(e), it will put you in an empty conference and > tell you the number. Presumably for anyone else to join the same > conference, you then have to tell them the number, e.g. by email, > IM or another phone call, and they then have to dial a different > extension which routes to MeetMe(without e). And if the empty > conference also has a PIN, does the first user need a list of > conference numbers to PINs so he can enter the correct PIN when > told the conference number? That's an administrative matter, not a detail of implementation. You could, of course, have the same PIN for multiple conferences. > This all seems rather cumbersome, and I haven't had the chance to > experiment with this feature yet, so the above probably highlights > both (a) my lack of understanding, and (b) the lack of > documentation! If the feature doesn't make any sense to you, then don't use it. For a customer of ours, though, it was necessary to have this feature. I would suggest actually trying out the feature a couple times, if your goal is to learn how to use it. -Tilghman ___ 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] Re: MeetMe 'i' option not working correctly?
On 03/09/06 16:41 Tony Mountifield said the following:
In article <[EMAIL PROTECTED]>,
Jon Webster <[EMAIL PROTECTED]> wrote:
I'm running 2.4.5 and app_meetme never plays conf-hasleft or
conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
determine the cause. Any suggestions are greatly appreciated.
exten => 600,1,MeetMe(600|i) I get the following:
-- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Mar 8 06:13:53 WARNING[5197]: channel.c:2535 ast_request: No channel
type registered for 'zap'
Mar 8 06:13:53 WARNING[5197]: app_meetme.c:461 build_conf: Unable to
open pseudo channel - trying device
The above messages indicate that chan_zap.so isn't loaded. Possibly it
isn't even built. You need to build *and install* zaptel before starting
to build Asterisk. Asterisk will find the zaptel libraries and will
build chan_zap.
MeetMe requires a timing device, you'd need either a zaptel line card or to
load ztdummy to provide pseudo timing.
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
+=+
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[asterisk-users] Re: Meetme... No channel type registered for 'zap'
In article <[EMAIL PROTECTED]>, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Kristian, > > I don't have any zap hardware What do I put in zaptel.conf if I don't > have any hardware? > On some other systems we have, with chan_zap not loaded, and no zaptel.conf > (running > 1.2.9.1), meetme runs fine. This system with the problem has 1.2.12.1. I > wonder if something > was changed? Doug, it sounds to me like you don't have the /dev/zap device files. Do you have the file /etc/udev/permissions.d/zaptel.permissions and /etc/udev/rules.d/zaptel.rules installed? What Linux distro are you using? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme... No channel type registered for'zap'
> -Original Message- > From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] > Sent: Wednesday, October 25, 2006 10:18 AM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Re: Meetme... No channel type registered > for'zap' > > > On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote: > > > -Original Message- > > > From: Tony Mountifield [mailto:[EMAIL PROTECTED] > > > Sent: Wednesday, October 25, 2006 1:26 AM > > > To: asterisk-users@lists.digium.com > > > Subject: [asterisk-users] Re: Meetme... No channel type > registered for > > > 'zap' > > > > > > > > > In article > > > <[EMAIL PROTECTED]>, > > > Douglas Garstang <[EMAIL PROTECTED]> wrote: > > > > Kristian, > > > > > > > > I don't have any zap hardware What do I put in > > > zaptel.conf if I don't have any hardware? > > > > On some other systems we have, with chan_zap not loaded, > > > and no zaptel.conf (running > > > > 1.2.9.1), meetme runs fine. This system with the problem > > > has 1.2.12.1. I wonder if something > > > > was changed? > > > > > > Doug, it sounds to me like you don't have the /dev/zap > device files. > > > > > > Do you have the file > /etc/udev/permissions.d/zaptel.permissions and > > > /etc/udev/rules.d/zaptel.rules installed? > > > > Tony, I don't have /etc/udev/permissions.d/, but I do have > the other file. > > > > demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules > > -r--r--r-- 1 root root 498 Oct 24 15:50 > /etc/udev/rules.d/zaptel.rules > > And its contents is? Contents are: demeter:(acd1)ipt # cat /etc/udev/rules.d/zaptel.rules # zaptel devices with ownership/permissions for running as non-root KERNEL=="zapctl", NAME="zap/ctl", OWNER="asterisk", GROUP="asterisk", MODE="0660" KERNEL=="zaptimer", NAME="zap/timer", OWNER="asterisk", GROUP="asterisk", MODE="0660" KERNEL=="zapchannel", NAME="zap/channel", OWNER="asterisk", GROUP="asterisk", MODE="0660" KERNEL=="zappseudo", NAME="zap/pseudo", OWNER="asterisk", GROUP="asterisk", MODE="0660" KERNEL=="zap[0-9]*", NAME="zap/%n", OWNER="asterisk", GROUP="asterisk", MODE="0660" > But do you actually have the channels? Anything in /dev/zap ? Anything > in /sys/class/zaptel ? Specifically pseudo/zapseudo . Do I have the channels? No, I don't think so. I don't have any zap hardware installed. That's why I am using ztdummy. demeter:(acd1)ipt # ls -l /dev/zap total 0 crw-rw 1 root root 196, 254 Oct 24 16:01 channel crw-rw 1 root root 196, 0 Oct 24 16:01 ctl crw-rw 1 root root 196, 255 Oct 24 16:01 pseudo crw-rw 1 root root 196, 253 Oct 24 16:01 timer demeter:(acd1)ipt # ls -l /sys/class/zaptel total 0 drwxr-xr-x 2 root root 0 Oct 24 16:01 zapchannel drwxr-xr-x 2 root root 0 Oct 24 16:01 zapctl drwxr-xr-x 2 root root 0 Oct 24 16:01 zappseudo drwxr-xr-x 2 root root 0 Oct 24 16:01 zaptimer 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] Re: Meetme... No channel type registered for 'zap'
In article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Tony Mountifield [mailto:[EMAIL PROTECTED] said:
> >
> > Doug, it sounds to me like you don't have the /dev/zap device files.
> >
> > Do you have the file /etc/udev/permissions.d/zaptel.permissions and
> > /etc/udev/rules.d/zaptel.rules installed?
>
> Tony, I don't have /etc/udev/permissions.d/, but I do have the other file.
>
> demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
> -r--r--r-- 1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules
>
> >
> > What Linux distro are you using?
>
> I'm using Gentoo Linux, and have been for a number of months. This is the
> first time this
> problem has cropped up. If I have ztdummy installed, why do I need the device
> files? Isn't
> that what ztdummy is supposed to do?
I'm not familiar with Gentoo, so I'm afraid I can only help in general
terms.
In fact I've gone back and re-read your original message and found that
I had misinterpreted it, so I'll start from the beginning again. It's
nothing to do with udev or device files after all.
The messages you mentioned were:
-- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack
-- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack
-- Playing 'conf-getconfno' (language 'en')
Warning, flexible rate not heavily tested!
Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type
registered for 'zap'
Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open
pseudo channel - trying device
-- Created MeetMe conference 1023 for conference '5000'
-- Playing 'conf-onlyperson' (language 'en')
-- Hungup 'IAX2/xxx.yyy.142.204:4569-2'
What you didn't say was whether the conference worked despite those
messages.
When you create a conference, MeetMe tries to create a full Asterisk
channel for the zaptel pseudo device. The two warnings above indicate
that it was unable to do so, meaning that chan_zap.so is not loaded.
If Meetme fails to create a full asterisk channel, it falls back to
opening a file descriptor on /dev/zap/pseudo directly. That's what the
"trying device" part in the second message means. It evidently
succeeded, or there would have been a third error message.
If conferences are working ok for you, you can ignore the warnings.
However, certain options such as 'i' will not work, as they rely on the
full Asterisk channel.
The best solution is to make sure that chan_zap was built when you
compiled Asterisk on this box, AND that you don't have an entry in
modules.conf preventing it being loaded ("noload=chan_zap.so").
To make sure chan_zap is built, you must have built AND installed zaptel
BEFORE you start to build Asterisk.
Hope this all helps!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Re: Meetme... No channel type registered for 'zap'
In article <[EMAIL PROTECTED]>, Douglas Garstang <[EMAIL PROTECTED]> wrote: > Tony, > > Thanks for the reply. chan_zap was built, but I am not loading it. The > meetme conference works, but user entry/exit is not being announced > (that's option 'i', right?). Yes, that's right. Also the 'r' option to record a conference won't work without chan_zap loaded, for the same reason. > I tried loading chan_zap, but it complains that I have no zaptel.conf > file. So, if I have no zap hardware, what should I put in zaptel.conf? I think you meant zapata.conf. The file /etc/zaptel.conf is required for the zaptel device modules, and /etc/asterisk/zapata.conf is for chan_zap. If you are using just ztdummy, then /etc/zaptel.conf can be used just as it comes in the zaptel distribution (zaptel.conf.sample), and the same is true with /etc/asterisk/zapata.conf - it can be taken straight from asterisk/configs/zapata.conf.sample and needs no changes. In fact if you do "make samples" in your asterisk directory, it will install default configuration files in the right place for you. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme... No channel type registered for 'zap'
In article <[EMAIL PROTECTED]>, Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 20:50, Wed 25 Oct 06, Tony Mountifield wrote: > > In fact if you do "make samples" in your asterisk directory, it will > > install default configuration files in the right place for you. > > do _NOT_ i repeat _NOT_ do this if you have your actual > configs in /etc/asterisk Unless you realise that the makefile copies your existing configs to backup files. You can then either copy the customised ones back, or use vimdiff to copy your customisations into the updated template. > It messed up my configs twice. Hmm, not nice :-( > "Why is it drug addicts and computer afficionados are both called users?" I think it comes from the verb "to use": drug addicts use drugs and computer afficionados use computers. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme... No channel type registered for 'zap'
In article <[EMAIL PROTECTED]>, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > I'm not having much luck here. I used the default zaptel.conf and zapata.conf > files, and put > a load => chan_zap.so in my modules.conf. On load, asterisk reports: > > [chan_zap.so] => (Zapata Telephony w/PRI) > Oct 26 08:24:33 ERROR[8419]: chan_zap.c:10147 setup_zap: Unable to load > config zapata.conf > Oct 26 08:24:33 WARNING[8419]: loader.c:414 __load_resource: chan_zap.so: > load_module > failed, returning -1 > Oct 26 08:24:33 WARNING[8419]: loader.c:499 load_modules: Loading module > chan_zap.so failed! > > and then it bombs out. :( OK, the warnings and the bombing out are just knock-on effects from the first error, which is telling you it can't read zapata.conf. It must be either a permissions problem or a format problem. Please post the output of the following commands: ls -ld /etc/asterisk ls -l /etc/asterisk/zapata.conf file /etc/asterisk/zapata.conf cat /etc/asterisk/asterisk.conf What command line are you using to start asterisk? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Re: Meetme and Sipura SPA-941 - bad jitter/distortion
"Ryan Booz" <[EMAIL PROTECTED]> writes: > Now, however, there is a (very) slight echo introduced into any calls made > to this extension. So obviously the way that the phone sends packets is > causing some issues. Anyone have a resource or guide to point me to on best > way to debug packet transmission for good calls? Are you sure the echo isn't acoustic echo from the handset itself? Its older sibling, the SPA-841 was really bad in this regard. On a purely sip call between two SPA-841's, if you bumped the earphone gain past halfway on the display the other side would invariably complain about the echo. I always wanted to fill the Sipura handset with modeling clay and see if that helped things any. (The echo was only a problem on direct sip-to-sip calls. Any calls going into the PSTN seemed to always be processed by an echo-can, so it wasn't noticed there.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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] Re: Meetme and Sipura SPA-941 - badjitter/distortion
It might be. I'm going to work with one of the remote users again tomorrow to see if we can get it working better. You're also right that the PSTN calls don't hear the echo, INSTEAD I hear a faint "static/waves on a beach" sound whenever I talk though a PSTN set through the system to this user. Pushing the packet size back to .03 makes direct calls better, but then MeetMe goes screwy again. ARG! :-) Anyone have experience with the mentioned fix at: http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call quality difference with SIP? I read the whole thing thinking it was going to end up saying this was a 1.2 feature, but looks like it got pushed to 1.3. Thoughts? Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S. Rupprecht Sent: Thursday, December 08, 2005 4:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Meetme and Sipura SPA-941 - badjitter/distortion "Ryan Booz" <[EMAIL PROTECTED]> writes: > Now, however, there is a (very) slight echo introduced into any calls made > to this extension. So obviously the way that the phone sends packets is > causing some issues. Anyone have a resource or guide to point me to on best > way to debug packet transmission for good calls? Are you sure the echo isn't acoustic echo from the handset itself? Its older sibling, the SPA-841 was really bad in this regard. On a purely sip call between two SPA-841's, if you bumped the earphone gain past halfway on the display the other side would invariably complain about the echo. I always wanted to fill the Sipura handset with modeling clay and see if that helped things any. (The echo was only a problem on direct sip-to-sip calls. Any calls going into the PSTN seemed to always be processed by an echo-can, so it wasn't noticed there.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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] Re: Meetme and Sipura SPA-941 -badjitter/distortion
> It might be. I'm going to work with one of the remote users again tomorrow > to see if we can get it working better. You're also right that the PSTN > calls don't hear the echo, INSTEAD I hear a faint "static/waves on a beach" > sound whenever I talk though a PSTN set through the system to this user. > Pushing the packet size back to .03 makes direct calls better, but then > MeetMe goes screwy again. ARG! :-) > Anyone have experience with the mentioned fix at: > http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call > quality difference with SIP? I read the whole thing thinking it was going > to end up saying this was a 1.2 feature, but looks like it got pushed to > 1.3. Thoughts? That patch and bug does help quite a few scenarios, but they won't help with this problem. MeetMe strictly assumes 20ms audio in 1.2.0. Earlier releases would and could process larger payloads, but the method used was identified as a source of increasing delay. The buffering used in 1.2.0 to send and receive audio packets from the zaptel mixing engine now drops anything past the initial 20ms. Check out http://bugs.digium.com/view.php?id=5697 for one possible fix. Dan ___ --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] Re: Meetme and Sipura SPA-941-badjitter/distortion
Dan, thank you for the pointer. I read through the whole thing and will potentially try this next week. I'll post back with any thoughts. Thanks! Ryan Booz Director of IT Good Steward Software, LLC 111 Sowers Street, Suite 400 State College, PA 16801 Phone: 877-327-3702 x.26 (814-237-3744 x.26) Fax: 719-623-0577 Visit us at www.energycap.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: Thursday, December 08, 2005 6:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: Meetme and Sipura SPA-941-badjitter/distortion > It might be. I'm going to work with one of the remote users again tomorrow > to see if we can get it working better. You're also right that the PSTN > calls don't hear the echo, INSTEAD I hear a faint "static/waves on a beach" > sound whenever I talk though a PSTN set through the system to this user. > Pushing the packet size back to .03 makes direct calls better, but then > MeetMe goes screwy again. ARG! :-) > Anyone have experience with the mentioned fix at: > http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2? Does it make call > quality difference with SIP? I read the whole thing thinking it was going > to end up saying this was a 1.2 feature, but looks like it got pushed to > 1.3. Thoughts? That patch and bug does help quite a few scenarios, but they won't help with this problem. MeetMe strictly assumes 20ms audio in 1.2.0. Earlier releases would and could process larger payloads, but the method used was identified as a source of increasing delay. The buffering used in 1.2.0 to send and receive audio packets from the zaptel mixing engine now drops anything past the initial 20ms. Check out http://bugs.digium.com/view.php?id=5697 for one possible fix. Dan ___ --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] Re: Meetme... No channel type registered for 'zap'
> -Original Message- > From: Tony Mountifield [mailto:[EMAIL PROTECTED] > Sent: Wednesday, October 25, 2006 1:26 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Meetme... No channel type registered for > 'zap' > > > In article > <[EMAIL PROTECTED]>, > Douglas Garstang <[EMAIL PROTECTED]> wrote: > > Kristian, > > > > I don't have any zap hardware What do I put in > zaptel.conf if I don't have any hardware? > > On some other systems we have, with chan_zap not loaded, > and no zaptel.conf (running > > 1.2.9.1), meetme runs fine. This system with the problem > has 1.2.12.1. I wonder if something > > was changed? > > Doug, it sounds to me like you don't have the /dev/zap device files. > > Do you have the file /etc/udev/permissions.d/zaptel.permissions and > /etc/udev/rules.d/zaptel.rules installed? Tony, I don't have /etc/udev/permissions.d/, but I do have the other file. demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules -r--r--r-- 1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules > > What Linux distro are you using? I'm using Gentoo Linux, and have been for a number of months. This is the first time this problem has cropped up. If I have ztdummy installed, why do I need the device files? Isn't that what ztdummy is supposed to do? 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
Re: [asterisk-users] Re: Meetme... No channel type registered for 'zap'
On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote: > > -Original Message- > > From: Tony Mountifield [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, October 25, 2006 1:26 AM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] Re: Meetme... No channel type registered for > > 'zap' > > > > > > In article > > <[EMAIL PROTECTED]>, > > Douglas Garstang <[EMAIL PROTECTED]> wrote: > > > Kristian, > > > > > > I don't have any zap hardware What do I put in > > zaptel.conf if I don't have any hardware? > > > On some other systems we have, with chan_zap not loaded, > > and no zaptel.conf (running > > > 1.2.9.1), meetme runs fine. This system with the problem > > has 1.2.12.1. I wonder if something > > > was changed? > > > > Doug, it sounds to me like you don't have the /dev/zap device files. > > > > Do you have the file /etc/udev/permissions.d/zaptel.permissions and > > /etc/udev/rules.d/zaptel.rules installed? > > Tony, I don't have /etc/udev/permissions.d/, but I do have the other file. > > demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules > -r--r--r-- 1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules And its contents is? But do you actually have the channels? Anything in /dev/zap ? Anything in /sys/class/zaptel ? Specifically pseudo/zapseudo . -- Tzafrir Cohen iax:[EMAIL PROTECTED]/tzafrir icq#16849755 mailto:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] http://www.xorcom.com ___ --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
