similar to: Problems with MeetMe accepting conference PIN

I ran in to an almost impossible ability to run MeetMe on our poweredge. Our PE is a 4 proc with OHCI and we had no need to buy any PCI cards other than for running MeetMe. If you know how to recompile your kernel, recompile it and make RTC a module (its compiled in by default). Then you can use zaprtc as your timing source. Be sure to `rmmod rtc` before you `insmod zaprtc` because zaprtc is a

I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I did a modprobe on ztdummy I was able to enter into a conference room using my softphone clients. I'm using SJphone and Firefly. I have noticed a significant delay (1 to 3 seconds) while talking within the conference room. I have tried both clients, SIP and IAX protocols and various codecs. I have also tried it from different

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

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 509@asterisk.mydomain.com (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

I haven't seen this appear on the list, so I thought I would resend it... Sorry for the repost if it did appear before... ----- Forwarded message from Michael George <george> ----- Date: Wed, 3 May 2006 21:48:09 -0400 From: Michael George <george> Subject: meetme conference latency degrades... To: asterisk-users@lists.digium.com We have recently started making more frequent use

i have problem to setup application meetme. i'm using asteisk-1.0.3 and sjphone as client. ++++++++++++++++++++++++++++++++++++ monchemin --------------------------------- Do you Yahoo!? Dress up your holiday email, Hollywood style. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi all, I'm configuring meetme in asterisk but no luck. I dont have a digium card, so based on the wiki, i have to install the zaptel driver and enable the ztdummy on it. I compiled, install the module and enable it on the module.conf. When i tried to start the asterisk, asterisk wont start because of the error in loading the meetme module. Jun 16 15:36:56 WARNING[-1084989312]:

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Hi all I want to use conference in Asterisk. I configure a conference room in meetme.conf (as conf => 600,1234) and extensions.conf as (exten => 600,1,MeetMe(600,i,1234)) . When i call the extension 600, i have the following message in the asterisk logs: WARNING[7758]: pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension (conference, 600, 1) == Spawn extension

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten => 7777,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials 7777 to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I

I am trying to use meetme() on SIP channels. I found this line on voip-info.org ------------- It *is* necessary either to have a Digium card or a dummy timing driver (e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that doesn't help you use AGI with SIP channels: They have no capacity to use any AGI script at all. If they try to, they get no audio. ----------------- I am

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

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

I'm trying to record a Meetme conference to disk, but the Monitor application doesn't seem to play nicely with Meetme. In extensions.conf, I have this: exten => 1000,1,Answer exten => 1000,2,Monitor exten => 1000,3,Meetme This starts up the monitoring OK, and it records the prompts that Meetme gives, but as soon as the user enters the conference, the -out WAV file stops

Hi all, Anyone know how to dynamically create meetme conference rooms on the flight? I remembered a while ago there was a switch that tell meetme to create the conference room is the room is not defined in the meetme.conf. It doen't seem to be working for me anymore. Thnx