similar to: conference authorization

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

Good day! Have a weird problem with HT-286 and Conference room. I use Asterisk CVS-HEAD-06/04/04. Here it is: When HT-286 get into the conference room first and nobody in that room everything seems ok (with any codec HT286 allowed), but when HT-286 get into conference room when somebody already there, have got different HT behavior: 1. When HT use GSM codec => it connects to conference room,

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

Hi all. I have come to the conclusion that there just isn't anything out there for allowing SIP and NAT to work together nicely. This is rather amazing considering that as far back as March 2000 there are documents describing how to do it. So I've started a really simple SIP and RTP proxy project, SaRP, on sourceforge.net. Yesterday we uploaded 0.2 of the perl based release. This is the

https://bugs.freedesktop.org/show_bug.cgi?id=36174 Summary: Xorg crashing in nv44 card in 3D apps Product: Mesa Version: 7.10 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

Howdie: I've been reading some old threads and still have a couple of questions about applying the AGI_BACKGROUND script inside a Conference. Perhaps someone can save me a bit of fidd'lin. Am I right in assuming that the MEETME_AGI_BACKGROUND script **WILL WORK** on SIP conferenced channels **WITHOUT** an **ACTIVE** zap channel-- AS LONG AS THERE IS A DIGIUM CARD INSTALLED IN THE

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Hi, The dialin conference via asterisk is over, one person is still in the conference room and accidentally does not hang up properly. Her meter at the phone company keeps running... I'd like to implement something to the effect of checking whether there is only one participant in the conference, and when this is the case, to cancel the call after a predefined time (perhaps 5 or 10 mins.

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 ? TIA Julian.

http://bugs.freedesktop.org/show_bug.cgi?id=23847 Summary: kernel BUG when using nouveau Product: xorg Version: 7.4 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy: shiningxc at

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

Dear all I want to display 4 dimensional space by some suitable way. I searched CRAN and found miscellaneous 3 dim graphics packages which I maybe can modify but anyway I am open to any hint how to efficiently display data like: longitude, latitude, height, value Thank you Petr Pikal petr.pikal at precheza.cz

Hello all ! I have a problem with my IVR with terminate connection from PSTN phone Here is my configuration extension.conf [ivri] ;exten => s,1,Wait(1) exten => s,1,Answer ;exten => s,2,DigitTimeout(5) ;exten => s,3,ResponseTimeout(10) exten => ivr,1,Background(demo-congrats) exten => 1,1,MP3player,/mnt/linux/mp3/song/04.mp3 exten =>

http://bugzilla.mindrot.org/show_bug.cgi?id=215 Summary: No warning for failed ssh -v -R Product: Portable OpenSSH Version: older versions Platform: UltraSparc OS/Version: Solaris Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: openssh-unix-dev at mindrot.org

Hi all. I'm trying to setup Asterisk to act as a purely SIP PBX for Internet based VoIP. I've got it configured with with a couple of users in sip.conf like so: [andrew] type=friend username=andrew secret=<secret> host=dynamic defaultip=192.168.26.21 dtmfmode=inband Calls addressed as 'sip:username@asterisk.server' or 'sip:extension@asterisk.server' work fine

Hi all, we observed a strange effect when copying an file within a samba share: Both atime an mtime of the target file are set to the mtime of the original file. The atime of the original file is updated to the current time. 1. Status of the original file: # stat test.txt File: `test.txt' Size: 3 Blocks: 8 IO Block: 4096 regular file Device: 811h/2065d

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP