similar to: SIP Ad-Hoc Conferencing with Asterisk

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

Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, " No ISA Tormeta

I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey

Hi. I am using asterisk-1.8 and I am having problems getting conferencing to work properly. I did modprobe on dahdi and did load => chan_dahdi.so in /etc/asterisk/modules.conf. Now I do get conferencing, but meetme says [Jan 15 11:38:56] WARNING[9214] app_meetme.c: No DAHDI channel available for conference, conference recording disabled (is chan_dahdi loaded?) Now chan_dahdi is indeed

Hello, We are building a conference server using a Dell PE 2850 3GHz with 2G memory. This conference server will be used to hold a large conference with 30-50 simultaneous users in a conference room. This large conference will take place two days per week for 3 hours each day. When a large conference is going on, no other conference room will be created. The rest of the week,

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

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 trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Hi, I am wondering whether I can use Asterisk as SIP/H.323 Signalling Gateway. The setup I envisage looks as follows: H.323 end-point ---------(ETH)--------- Asterisk ---------(ETH)--------- SIP Proxy/Registrar ---------(ETH)--------- SIP end-point (ETH: Ethernet) In principle, Asterisk would just be used to integrate H.323 end-points into a fully SIP-based core-network. Hence, there

My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but

Any one knows how to make Meet Me video conferencing room. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

Dear list, we would really like to be able to "invite a third and fourth party" to our current one-on-one call. At the moment, we have to agree to dial into MeetMe 10 minutes later, then make calls to the third parties, and hope it all works out. I have found a couple of examples on the Internet for converting channels into conferences, but I could not get any of them working. Does

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

Hello! I'm new in Asterisk and I hope that my trouble is very simple. We're implementing a Education Project of a e-Learning system (LMS) that uses conferencing (video and audio) over internet. The e-Learning system will be on GPL license, and for that, we're using only free software to implement. Asterisk is our first choice for video and audio conferencing, and making tests,

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will