search for: extrachannel

...d: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out channel: SIP/sip-gate/0394839405 --------------------------------- Then talk to each other for a while... --------------------------------- action: redirect priority: 1 exten: 1234 context: conference channel: SIP/sip-gate-0868b000 extrachannel: SIP/sip-gate-086a5000 action: logoff --------------------------------- This approach works but results in a bad sound quality after the redirect. The sound seems to be scrambled. Before redirecting the sound quality is quite well, of course. All extensions are called via SIP with the same cod...

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

...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 putting them both in the same Meetme conference. What I want to do is find a way to take two unrelated existing channels (which for the sake of argument might be sitting in MusicOnHold, separate conferences, the same conference or whatever), and link them to...

Doe anybody know of a way to redirect both channels from a bridge to different dial plan extensions from the using the AMI. Currently, as soon as I redirect one of the channels the other appears to be dropped and gets reorder tone (congestion, fast busy). I guess what I really need is a way to redirect one of the channels and hold on to the other. Thanks, Neil Cherry

...much appreciated. I would like to use the manager API to take an existing call on a specific SIP extension, dial and conference in a third party. From what I can tell, the way to do this would be to take the two original parties on the call and stick them in a meetme room using Redirect with ExtraChannel, then dial the new party and also dump them into the meetme room. The problem I am having is this: I know the extension of the SIP phone that is on the call, but I don't know it's channel, or the channel of the other party. I need to figure both of these out to be able to use the Man...

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can

...sf.net this is my problem: i will grant to users the possibility to transfer calls to other users using a web interface, for example if SIP/200 is talking with SIP/400 who wants to transfer the call to SIP/500 i use this commands with manager API: Action: Redirect\r\n Channel: SIP/200-sads\r\n ExtraChannel: 500\r\n Exten: 500\r\n Context: from-internal\r\n Priority: 1\r\n\r\n this works fine (maybe the sintax now isn't correct... but it works), but my problem is that the call is immediately transferred to 500. I'd like if: 1 - 200 calls 400 2 - 400 want to transfer the call to 500 3 - 400...

...nnels. 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: SIP/voip_out_22-809c Exten: 111 Context: meetme-test Priority: 1 Then , the channel named SIP/voip_out_22-809c has been transfered to the conference 111. But, the channel named SIP/612-5456 has been hangup automatic. The context meetme-test is : [meetme-test] ex...

...connect each one with the final extension. You didn't tell, which version of asterisk You are using. In 11 and later there is the new conference module, which makes it easier. In the first step You can use AMI REDIRECT to transfer both parties into one dynamic conference. Use the Channel and ExtraChannel to take both channels. In the second step use AMI Join Events to trigger your next transfer to the different extensions in Your dialplan. Each channel joining the conference will generate a separate event. HTH, Karsten

I'm very interested in the redirect feature of Asterisk. So far I haven't gotten it to work. My scenario is that there is a two party call going on where I want to send one of those parties somewhere else. In the wiki is only an example how to send both parties to a meetme room. Is the ExtraChannel parameter required? This is what I have: Action: Redirect Channel: SIP/8080-e2a7 Exten: 5000 Context: local Priority: 1 Now, the SIP/8080 channel is connected to one extension, if I now redirect it to extension 5000 (as shown above), the SIP channel hears a short "ring" tone and the...

Hi, BRIstuff contains two bugs in its implementation of the Redirect manager action: 1. If the property ExtraUnqiueId is used, the Priority property is used to redirect the extra channel (instead of ExtraPriority) 2. If the property ExtraChannel is used, 0 is used to redirect the extra channel regardless of the Priority and ExtraPriority properties. A patch for manager.c is available at http://www.reucon.net/~srt/bristuff_redirect.patch as a result to a bug filed against Asterisk-Java at http://jira.reucon.org/browse/AJ-34 I've sent...

I am looking for a way to have an agent execute an attended transfer using the asterisk manager interface (AMI). I have been trying to use the dual Redirect from svn trunk. The problem with this function is that the "ExtraChannel" does not get redirected properly afaict. Now, I am looking for other solutions for the list, I will probably try playing DTMFs on the agent channel to simulate the manual transfer next unless anyone has some better ideas. Thanks Grant

Check this out. I recently closed a bug I had written, #495 "ExtraChannel in transfer causes crash" Now I've been able to reproduce it, and somewhat narrowed down the culprit. But before I write another bug report, I wanted to see if anyone else had experienced the following (or would like to try:) When DNS (or outside connection to the network, not sure w...

Is it possible to park a call through the manager interface? If yes; how? Regards Thorben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/ec2a5f90/attachment.htm