similar to: Dropped calls if transferred across servers into MeetMe with mobile source

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

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

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-----------------------------------------:

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, 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 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 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

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,

Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. The IaxClient seems to use silence suppression, and not sure if this can be disabled. The client works fine through most routers, but for some it disconnects the client after about 5 minutes and gives this error in the asterisk logs: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to

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

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

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>

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]:

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- 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

I was searching for help on how to handle the errors that are returned from the MeetMe application. for instance. 1) if a user tries to join a conference that is locked, allison says that the conference is locked and then comes back to the dialplan, however it does not continue down the dialplan. I have a meetme command on Priority 8, and the CLI says that it returned non zero (as the wiki

hi, i write this looking for free conference room, i checl code and don?t see any error but die at priority 7 if room 1001 have users in exten => _1NXXNXXXXXX,1,RouteCall(${EXTEN}) exten => _1NXXNXXXXXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13) exten => _1NXXNXXXXXX,3,Setvar,var=0 exten => _1NXXNXXXXXX,4,MeetMeCount(1001|var) exten => _1NXXNXXXXXX,5,GotoIf($[${var} =0]?7:6)

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,

This message was just bounced back to me. I am not sure if it made it to the list originally or not, as I received no responses. Since this message was written, I have installed Zap hardware into this server. The Zap channels can be transferred to the Meetme conference. The IAX2 calls still cannot. Any suggestions will be greatly appreciated. Sincerely, Trevor Hammonds Trevor G.

Hi again, I have search a bit more in order to discriminate the problem, and I have realized that everybody goes out from a conference room, when a mobile phone o SIP call press '#'. Instead when a analog phone press '#', only this phone goes out. I'm working in Spain using a E400P, so I don't know if it's a zaptel configuration problem, an * version problem, a

Hi everybody, I'm trying to use MeetMe(2000|p) in order to enter in a conference room. But when a mobile or a SIP call press '#' to go out from it, everybody goes out. Instead when a analg press it, all works fine. Anyone else have this problem?? Can anybody help me please?? Thanks a lot. Regards.