similar to: bad sound quality after Redirect

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

Hello! I am having problems figuring out how to do something, and any help would be 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,

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

i am developing a web application to manage callcenter, i will shortly release it on 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:

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:

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

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

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

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

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

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

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: > 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

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

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 which) is broken and

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.