similar to: Asterisk SIP Channels Bridge

Hi there, I'm trying to bridge 2 SIP channels together via AGI script. The AGI script is written in C#. The first caller would call in and be placed on hold and the second caller would call in and both the calls gets connected together. But I am having problem with the second caller finding the first channel. Can someone point me to the right direction? thanks Eric

Hi everybody. I am in the following scenario: 1 Customer "A" calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The AGI script originates a calls to a stand-by operator's cell phone (operator

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

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 have been successful using ChanSpy on a standard Dial call but when attempting to ChanSpy on an incoming call that has been added to a MeetMe conference (attempting to coach an agent that is speaking to a conference of callers) it seems to fail to connect to the channel. Here's the console dump: -- Accepting call from '2154182700' to '3399' on channel 0/18, span 4

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

Hello-- What I'd like to do: Use IAX softphones running on computers, in Auto-answer mode, with sound going to speakers, as a sort of public announcement system. What isn't working: Well, my first experiment was to set up the MeetMe system described on the Wiki... This works fine for voice announcements. You pick up a phone, dial the right extension, and an agi is fired up to put files

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

I've had a quick look through the mail list and wiki but haven't yet resorted to looking at the meetme source code.. I see references to a background agi script that can run if you're using Zap channels. Am I right in saying that that script runs for each channel in the conference? Or is it a one time deal, running when the conference is created? The backgrounder behind my question is

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

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

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

There is one more thing to try: http://snapvoip.blogspot.com/2009/07/appkonference-asterikast-high.html I would appreciate if anyone can comment on the feasibility of playing an audio file to the caller and callee using ControlPlayBack and appkonference. Much of the reviews indicate that appkonference is an over-kill for an audio as its main functionality is with video. Going past that. Thanks

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

I'm having trouble getting an AGI exec command to spawn app_disa. The script executes properly, but does not spawn DISA. The CLI gives no helpful clues. Am I doing the exec incorrectly? I want to have a way to authenticate callers to the extension by Caller ID... if their caller ID is in my database and set to active, they can call out. [like a calling card but auth'd by CID instead

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes