similar to: Grandstream HT286 1.0.4.63 & Meetme

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

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

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

I am using asterisk Version: 1:1.4.11~dfsg-1 as found in Debian. I'm using a call file to connect a meetme conference to an AGI script which plays files using the stream_file method. I have four files which should play in sequence, though only the first two files actually play. I get these errors in the CLI: [Sep 16 06:20:43] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 276 Buffer

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

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>

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

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.

I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays "incoming call" on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried different cables but some

I've dealing with this issue for a while, and I'd really like to know if anybody has experienced the same pain before :-) I've a lot of Grandstream HandyTone 286, loaded with the latest firmware (1.0.8.16) from the GS website. In my sip.conf, this ATA's are configured as: [05] type=friend username=05 secret=XXXX callerid="User 05" host=dynamic nat=yes qualify=yes

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

I have an application where calls come into an *box from a DID provider, and may be transferred to a meetme conference on another *box (the call is released by the first *box after transfer). These are ulaw IAX channel calls, and if the source is from a Verizon or Nextel mobile phone to the DID (other carriers not tested), the call drops about 2-3 minutes after it joined the meetme

Hi there, One of our Asterisk 1.2 machine is experiencing problems with MeetMe. Whenever meetme runs, the console is flooded with warning messages: The messages started as "No such file or directory" and becomes "Resource temporarily unavailable". I couldn't figure out what file MeetMe might be looking for, could anyone help? May 4 08:57:38 WARNING[19032]:

Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf => 6000 In my extensions.conf file I have: exten => 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me "this is not a valid conference number". Can anybody telling me what I'm doing wrong here?

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.

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

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]: