similar to: how to bridge two channels ?

Hello I'd like to know what my options are to bridge two channels after calling each through Dial(). I know about MeetMe, Conference, and Konference. Are there other options available just to bridge two calls? www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe www.voip-info.org/wiki/view/Asterisk+cmd+Conference www.voip-info.org/wiki/view/Asterisk+cmd+Konference I'd like the simplest

I would like to play audio--using controlplayback-- to 2 channels--agent and caller- simultaneously. Tried meetme,confbridge,originate without success. Tried redirecting the channels to a context, playing audio to the agent's channel and then bridging the 2 channels. The problem with this is as soon as the bridge is created the audio stops. I can provide the dialplan details, if anyone is

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for

One thing that concerns me is that this post is from 2009, even though the newest version of the app on Sourceforge is up to date. I have a customer who has been using a conference server that I built for him using app_konference for several years now and he routinely runs conferences with anywhere from 10 ? 125 active users. The ultimate goal is several hundred concurrent users and I can see that

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

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

Is there a way in Asterisk (I guess there's, it's only I can't figure out how :-)) to: 1.- Generate a call to channel 1 (example, to PSTN v?a an E1 card, using Zap/g1) 2.- Generate a call to channel 2 (example, an internal SIP extension). 3.- Once both channel have answered, connect the call between them. This way, I can, for example, play audios in both channels before they are

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hello everybody, using the manager api (via asterisk-java) I originate a call with application MeetMe to some extension (IAX). The agent joins the meetMe room on answering that incoming call. So far so good. Now I'd like to redirect that agent from the meetMe room to another meetMe room *only by using the manager api*. Is that idea possible to realize? Or has the agent to be involved?

Hello, Perhaps i'm wrong but i don't find a real documentation for cmd Bridge (i've take a look to the source code but i'm not a guru and i probabely miss something): Is it possible as for cmd meetme to have a context to return on 'exit'/end of the bridge? (in fact i think 'no') I've done a workaround with meetme but i would use Bridge (or confbridge) if

Hello, After battling with the Asterisk Manager interface(and getting it to pretty much do everything I want to do with it) I thought I'd share my experiences with those who are developing or are thinking of developing applications using it. First here's a list of some of the things the manager interface will let you do: - Dial a call from any extension/resource to any other

Hi all, I'm planning to develop a custom autodialer application which will be dealing with its own model for agents and queues, therefore it won't use neither asterisk agents nor asterisk queues, nor asterisk cdr. The application will supply the whole reporting and agent managing features by itself. The application will command asterisk through an AMI telnet connection using only the

Anyone have a suggestion on how I can have my asterisk box make two SIP or IAX calls and bridge the two together? It seems like it would be easy to setup but the only way I'm finding seems to be setting up meetme rooms. Jess

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel:

Apparently there are subtle diferences between meaning of MeetmeJoin event depending on channel type. Problem is: after originating a call from channel to MeetMe room i.e.: [meetme] exten => 1,1,Answer exten => 1,2,Meetme(kolejka|dqM) than: Context: meetme Exten: 1 Priority: 1 ActionID: 1077925740-00000004 Timeout: 5000 Action: Originate Async: true Channel: somechannel I get eventually

I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/conf@default Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of

Hello, When using xen without a bridge but NAT or routing, HVM domains can''t boot, and qemu-dm-n.log contains: config qemu network with xen bridge for tap0 xenbr0 bridge xenbr0 does not exist! That''s because the qemu-ifup script always tries to add the vif to a default-named xenbr0 bridge. On the contrary, PV domains just work fine with the same configuration file except HVM

> > 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

Hello everyone, I'm trying to bridge 2 SIP channels together via AGI script. The AGI Script is written in C#. I'm able to get the unique name of the channel and insert that into my database and need to know if i need to do anything in the dialplan so i can run EXEC in the agi script. The first caller would call in and be placed on hold and the second caller would call in and both the

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 =>