Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis" <panny@hotlinks.co.uk> To: <asterisk-users@lists.digium.com> Sent: Wednesday, October 22, 2003 1:43 PM Subject: [Asterisk-Users] Meetme> Is app_meetme broken? > > I seem to get invalid conference number all the time :( > > Panny > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
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... Tim -->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << >> tps@buoy.com >< (631) 399-2910 (888) 924-3728 << >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
Hi people, I have a user that forgets to hangup his conference calls, so they go on forever. Is there a way of limiting the duration of a conf call? Thanks in advance, Pablo -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199
Pablo Endres wrote:> Hi people, > > I have a user that forgets to hangup his conference calls, so they go > on forever. Is there a way of limiting the duration of a conf call? > > Thanks in advance, > > PabloTry using ABSOLUTETIMEOUT before starting the conference?
*CLI> help application AbsoluteTimeout
I am trying to get a simple MeetMe running between a few SIP phones here in our
office.
Here is a clip from my extensions.conf
exten => 9000,1,Ringing
exten => 9000,2,Answer
exten => 9000,3,Wait(1)
exten => 9000,4,MeetMe(|Md)
Here is what my console says:
-- Executing Ringing("SIP/loni-b550", "") in new stack
-- Executing Answer("SIP/loni-b550", "") in new stack
-- Executing Wait("SIP/loni-b550", "1") in new stack
Aug 6 13:48:56 WARNING[-298230864]: pbx.c:1257 pbx_extension_helper: No
application 'MeetMe' for extension (from-sip, 9000, 4)
== Spawn extension (from-sip, 9000, 4) exited non-zero on
'SIP/loni-b550'
My meetme.conf file looks correct (nothing really in it, just a [rooms] block).
Perhaps meetme isnt installed? I used an RPM for RC-1 I found on a mirror.
--
Travis Conway
travisconway@charter.net
FWD: 414668
+1 334 220-7519 (T-Mobile)
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I have just started using * and have been trying to set up MeetMe. So far I have not been able to start a conference. When I dial the conference extension (I am using X-Lite softphone), the call hangs up. I am not using Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My configuration is as follows: In meetme.conf: conf => 1234 In extensions.conf: exten => 2002,1,Answer exten => 2002,2,Wait(1) exten => 2002,3,MeetMe,1234 exten => 2002,4,Hangup At the CLI prompt I get the following: Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No application 'MeetMe' for extension (sip, 2002, 3) Is there anything I am missing? Any help will be greatly appreciated. Regards, Mitul
Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru.> I have just started using * and have been trying to set up MeetMe. Sofar I> have not been able to start a conference. When I dial the conference > extension (I am using X-Lite softphone), the call hangs up. I am notusing> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My > configuration is as follows: > > In meetme.conf: > conf => 1234 > > In extensions.conf: > > exten => 2002,1,Answer > exten => 2002,2,Wait(1) > exten => 2002,3,MeetMe,1234 > exten => 2002,4,Hangup > > At the CLI prompt I get the following: > Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No > application 'MeetMe' for extension (sip, 2002, 3) > > Is there anything I am missing? Any help will be greatly appreciated. > > Regards, > Mitul > > _______________________________________________ > 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 > >Thanks & Regards, Gurdeep (Guru) +91-11-35372111 guru@baggafamily.net
Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru.> I have just started using * and have been trying to set up MeetMe. Sofar I> have not been able to start a conference. When I dial the conference > extension (I am using X-Lite softphone), the call hangs up. I am notusing> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My > configuration is as follows: > > In meetme.conf: > conf => 1234 > > In extensions.conf: > > exten => 2002,1,Answer > exten => 2002,2,Wait(1) > exten => 2002,3,MeetMe,1234 > exten => 2002,4,Hangup > > At the CLI prompt I get the following: > Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No > application 'MeetMe' for extension (sip, 2002, 3) > > Is there anything I am missing? Any help will be greatly appreciated. > > Regards, > Mitul > > _______________________________________________ > 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 > >Thanks & Regards, Gurdeep (Guru) +91-11-35372111 guru@baggafamily.net
Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru.> I have just started using * and have been trying to set up MeetMe. Sofar I> have not been able to start a conference. When I dial the conference > extension (I am using X-Lite softphone), the call hangs up. I am notusing> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My > configuration is as follows: > > In meetme.conf: > conf => 1234 > > In extensions.conf: > > exten => 2002,1,Answer > exten => 2002,2,Wait(1) > exten => 2002,3,MeetMe,1234 > exten => 2002,4,Hangup > > At the CLI prompt I get the following: > Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No > application 'MeetMe' for extension (sip, 2002, 3) > > Is there anything I am missing? Any help will be greatly appreciated. > > Regards, > Mitul > > _______________________________________________ > 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 > >Thanks & Regards, Gurdeep (Guru) +91-11-35372111 guru@baggafamily.net
Hi, You need ztdummy. Follow this link, it worked for me, http://www.voip-info.org/wiki-Asterisk+timer+ztdummy Thanks, Guru.> I have just started using * and have been trying to set up MeetMe. Sofar I> have not been able to start a conference. When I dial the conference > extension (I am using X-Lite softphone), the call hangs up. I am notusing> Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My > configuration is as follows: > > In meetme.conf: > conf => 1234 > > In extensions.conf: > > exten => 2002,1,Answer > exten => 2002,2,Wait(1) > exten => 2002,3,MeetMe,1234 > exten => 2002,4,Hangup > > At the CLI prompt I get the following: > Aug 16 15:19:43 WARNING[565265]: pbx.c:1257 pbx_extension_helper: No > application 'MeetMe' for extension (sip, 2002, 3) > > Is there anything I am missing? Any help will be greatly appreciated. > > Regards, > Mitul > > _______________________________________________ > 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 > >Thanks & Regards, Gurdeep (Guru) +91-11-35372111 guru@baggafamily.net
I'm presently using meetme extensively on my server. I have a rather strange question. I'm using it with one person in "talk-only" mode and everybody else in "monitor" mode. I'm running an athlon xp 2800 with 1gb of ram. I can handle 40 users. Does anybody know of any adjustments that could be made to the asterisk to increase this limit? Or other changes. I can buy a newer system but I'm not sure how much that will help. I'm using only u-law presently. Is that a good thing or bad? Darren Wiebe darren@hagenhomes.com
I've got a problem with MeetMe. I dial the extension that dynamically
creates the new conf, but it just hangs up on me after telling me I'm
the only person in the conference. Here's my extensions.conf and what
its doing:
-- Executing Answer("SIP/101-74c0", "") in new stack
-- Executing Wait("SIP/101-74c0", "1") in new stack
-- Executing MeetMe("SIP/101-74c0", "|Dx") in new stack
-- Playing 'conf-getconfno' (language 'en')
-- Playing 'conf-getpin' (language 'en')
-- Created MeetMe conference 1023 for conference '100'
-- Playing 'conf-onlyperson' (language 'en')
-- Hungup 'Zap/pseudo-874077465'
== Spawn extension (default, 800, 3) exited non-zero on 'SIP/101-74c0'
[meetme-int]
exten => 800,1,Answer
exten => 800,2,Wait(1)
exten => 800,3,MeetMe(|Dx)
Tim Jackson
Network Engineer
Angelina County, Texas
(936)639-4827x101 office
(936)414-6723 mobile
I figured it out. I don't need the 'x' in there. 'x' - close the conference and hangup on all others when last marked user exits I'm an idiot, sorry. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kristian Kielhofner Sent: Wednesday, October 27, 2004 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MeetMe Tim Jackson wrote:> I've got a problem with MeetMe. I dial the extension that dynamically > creates the new conf, but it just hangs up on me after telling me I'm > the only person in the conference. Here's my extensions.conf and what > its doing: > > > -- Executing Answer("SIP/101-74c0", "") in new stack > -- Executing Wait("SIP/101-74c0", "1") in new stack > -- Executing MeetMe("SIP/101-74c0", "|Dx") in new stack > -- Playing 'conf-getconfno' (language 'en') > -- Playing 'conf-getpin' (language 'en') > -- Created MeetMe conference 1023 for conference '100' > -- Playing 'conf-onlyperson' (language 'en') > -- Hungup 'Zap/pseudo-874077465' > == Spawn extension (default, 800, 3) exited non-zero on'SIP/101-74c0'> > [meetme-int] > exten => 800,1,Answer > exten => 800,2,Wait(1) > exten => 800,3,MeetMe(|Dx) > > > Tim Jackson > Network Engineer > Angelina County, Texas > (936)639-4827x101 office > (936)414-6723 mobile >Tim, Even though the conference is dynamic, (I think) you still need to define a room number: exten => 800,3,MeetMe(800|Dx) Try that and let us know. -- Kristian Kielhofner _______________________________________________ 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
I know it's on the wiki somewhere, but I've been searching for 2 days on the wiki and google. I just can't seem to find the web interface for the meetme application. Does anyone have a link I could use. I found ASGI's dynamic conferences which is neat. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d4e94b9a/attachment.htm
>Can you please not post html. I can't correctly reply to this message.Sorry about the HTML, I had to reload my windoze machine and I forgot to turn it off and turn plain text on.>>I know it's on the wiki somewhere, but I've been searching for 2 days onthe wiki and google. I just can't seem to find the web interface for the meetme application. Does anyone have a link I could use. I found ASGI's dynamic conferences which is neat. Thanks in advance.>I think the app you are thinking of is meetme2.http://www.areski.net/asterisk-meetme/about.php?s=0 Thanks. For the reply -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/fbc17548/attachment.htm
Can it be that the MeetMe application is not installed by default even if there is a meetme.conf ? pbx.c:1280 pbx_extension_helper: No application 'MeetMe' for extension (from-sip, 550, 4) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050102/a8c15775/attachment.htm
Hi,
Can someone see what's wrong here please ?
I've installed the ztdummy driver to enable meetme, put his in my
extension.conf
exten => 550,1,Answer
exten => 550,2,Wait(1)
exten => 550,4,MeetMe(18|Md)
exten => 550,5,Hangup
this in my meetme.conf
[rooms]
;
; Usage is conf => confno[,pin]
;
conf => 18
and when I call 550 I get this error and the MusicOnHold (exten =>
550,4,MeetMe(18|Md)) also doesn't work:
-- Executing Answer("SIP/ses-0730", "") in new stack
-- Executing Wait("SIP/ses-0730", "1") in new stack
Jan 3 01:40:38 WARNING[12000]: pbx.c:1934 ast_pbx_run: Timeout, but no
rule 't' in context 'from-sip'
Thnx,
Hey All, Just finished installing Asterisk and configured all the necessary parameters to start. I can't seem to find the Meetme application in my asterisk directory. I downloaded asterisk from CVS and installed it and all my Snom phones are working and voicemail too. I am getting error: - Feb 11 17:10:19 WARNING[13042]: pbx.c:1280 pbx_extension_helper: No application 'Meetme' for extension (sip, 5557, 1) == Spawn extension (sip, 5557, 1) exited non-zero on 'SIP/phone1-f88d' Do I need zaptel to be installed? Any help will be appreciated. Nitesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050211/6f16242e/attachment.htm
Hi, I just recently installed Asterisk 1.0.7 but I cannot figure out how to install the MeetMe application. I don't think it installed with the standard 'make install' command. If not, how do I accomplish this? Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050417/4b6f2e1b/attachment.htm
Anyone notice on latest SVN trunk that meetme no longer asks for a pin when you try to enter a room? I want to verify it before I post it as a bug. John Bittner Simlab.net
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, 9999, 2) == Spawn extension (internal, 9999, 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extension&hl=en&gl=us&ct=clnk&cd=4&lr=lang_en&client=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles
I'm having issues getting meetme to work: Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (internal, 9999, 2) == Spawn extension (internal, 9999, 2) exited non-zero on 'SIP/mileslap-569b' the only thing I could find was this: http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extension&hl=en&gl=us&ct=clnk&cd=4&lr=lang_en&client=firefox-a but I have the timer working (I think): lsmod | grep dummy ztdummy 2608 - I'm really confused as to what to do next, if someone could help me out that would be great: I'm using gentoo with kernel 2.6.15. asterisk has been compiled from scratch with asterisk 1.2.5(I know not the latest) and zaptel 1.2.5 Thanks Miles
Snip..> >>> Thanks > >>> > >>> Miles > >> > >> If you type "modprobe zaptel" "modprobe ztdummy" at the Linux CLI, > >> what do you get? > > Nothing, they were loaded before, and loaded just fine. > > > > lsmod Module Size Used by > > ztdummy 2608 - > > rtc 10620 - > > zaptel 186468 - > > crc_ccitt 1576 - > > 3c59x 40240 - > > _______________________________________________ > > > And your dialplan for extension 9999? >Also post a 'show applications' form your asterisk CLI> prompt.>
Snip..> > you could try dialing 9999 from another phone and to dial > either 1023 > > or 0, my guess is 1023 is what the other people will have to dial. > I would assume that it would work like that, but nope. 9999 > from a different phone just creates a new conf, and 1023 is > never announced it is only in in the logs. > > where would a person find out about how meetme() works?>From the CLI you can do show application meetme.I would suggest you add a conference number to the extension. IE Exten => 750,1,MeetMe(750| your options go here ) Exten => 751,1,MeetMe(750| your options go here ) Exten => 752,1,MeetMe(750| your options go here ) Exten => 753,1,MeetMe(750| your options go here ) That will give you 4 meetme rooms. Play with that and then add and remove the different options that are in MeetMe to 'tweak' your install. Alex
Hi Everyone, Things seem to work fine (except my phone audio issue in a previous mail) I can leave a vmail message and it emails it out fine. However when I dial the vmail server from any phone it usually resets the phone half way through. There is no single point where it starts to do this, it can vary but it happens sometimes after I connect to the vmail server. Has anyone seen this? What other details can I post? Thanks Paul
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? Thanks in advance! Giuseppe
Just reposting as I know a lot of fresh faces are online :-) Any help appreciated Thanks Paul ----- Original Message ----- From: "Paul A Brown" <paul@fowlmere.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, April 09, 2006 8:44 PM Subject: [Asterisk-Users] Problem - Voicemail resets phone> Hi Everyone, > > Things seem to work fine (except my phone audio issue in a previous mail) > > I can leave a vmail message and it emails it out fine. However when I dial > the vmail server from any phone it usually resets the phone half way > through. There is no single point where it starts to do this, it can vary > but it happens sometimes after I connect to the vmail server. > > Has anyone seen this? What other details can I post? > > Thanks > > Paul > _______________________________________________ > --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 >
Anyone any ideas? They are SIP phones. I am not sure if its an asterisk or phone problem. Any help to isolate would be good. Thanks Paul ----- Original Message ----- From: "Paul A Brown" <paul@fowlmere.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Sunday, April 09, 2006 8:41 PM Subject: [Asterisk-Users] Cisco 7960 problems> Hi All, > > Not sure if this is a phone problem or an Asterisk problem. > > Basically after a period of time (around 30 minutes but not too sure of > the time) the phone no longer delivers any sounds. What I mean by that > is..... > > if I pick up the phone after a reset I get a dialtone. After around 30 > minutes and I pick up phone I get no dial tone but I can still dial. I > dialled the voicemail number, I can see on the asterisk console its asking > for which vmail box and password but I hear nothing. Anyone heard anything > like this before? > > Thanks > > Paul > _______________________________________________ > --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 >
Hi List,
I have the following in my extensions.conf. For some reason if the user enters
a room that does not exist instead of going to the next pri. it just says room
invalid and dumps the call. Can it be a bug ?
Exten => _*5XXX,1,MeetMe(${EXTEN:1},D)
Exten => _5XXX,1,MeetMe(${EXTEN},cMrpsq)
Exten => _5XXX,2,Goto(MainIVR,s,1)
I also tried addding the following which didnt work:
Exten => _5XXX,102,Goto(MainIVR,s,1)
Thanks a lot.
Dovid
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