asterisk users - Dec 2004 - API manager - Redirect with ExtraChannel

I am reading the WiKi page for the The Asterisk Manager API and I am 
interested in the functionality provided by the Redirect with 
ExtraChannel event.  There is minimal explaination on the WiKi page. Can 
anyone help/direct me to more doc on the subject.

regards
moe smadi

We use this in the astGUIclient to transfer an active conversation(both
parties) to a meetme room:

Action: Redirect
Channel: Zap/73-1
ExtraChannel: SIP/199testphone-1f3c
Exten: 8600029
Context: default
Priority: 1


where 8600029 is a meetme room.

Works very well.

Sadly like most obscure features in Asterisk it is not documented anywhere
very well. But ExtraChannel in Redirect is the only way to send both parties
on a 2-party call into a meetme room so that they can be joined by a 3rd
party(without having a multi-line phone that is).

MATT---



-----Original Message-----
From: m. smadi [mailto:smadim2@grads.ece.mcmaster.ca]
Sent: Monday, December 27, 2004 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] API manager - Redirect with ExtraChannel


I am reading the WiKi page for the The Asterisk Manager API and I am 
interested in the functionality provided by the Redirect with 
ExtraChannel event.  There is minimal explaination on the WiKi page. Can 
anyone help/direct me to more doc on the subject.

regards
moe smadi
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Hi,
We try to do something like somone did in redirect  API, but not fully
success...

This is what we did,  Both channel has been setup and talking...

Action: Redirect
Channel: SIP/210.201.75.100-081b9170
ExtraChannel: SIP/route886x-79cb
Exten:18
Context:sip
Priority:1

I have two issue:
1. Channel and Extrachannel could be the same tech channel, sip?
2.  Always one certain party connected, one disconnect //Zombie ?? why??


Event: Link
Channel1: SIP/210.201.75.100-08168dd0
Channel2: SIP/route886x-5550
Uniqueid1: 1108739916.2
Uniqueid2: 1108739925.3

Action: Redirect
Channel: SIP/210.201.75.100-08168dd0
ExtraChannel: SIP/route886x-5550
Exten: 18
Context: sip
Priority: 1

Event: Newchannel
Channel: AsyncGoto/SIP/route886x-5550
State: Up
Callerid: <unknown>
Uniqueid: 1108739972.4

Event: Rename
Oldname: SIP/route886x-5550
Newname: SIP/route886x-5550<MASQ>
Uniqueid: 1108739925.3

Event: Rename
Oldname: AsyncGoto/SIP/route886x-5550
Newname: SIP/route886x-5550
Uniqueid: 1108739972.4

Event: Rename
Oldname: SIP/route886x-5550<MASQ>
Newname: AsyncGoto/SIP/route886x-5550<ZOMBIE>
Uniqueid: 1108739925.3

Event: Newexten
Channel: SIP/route886x-5550
Context: sip
Extension: 18
Priority: 1
Application: Answer
AppData:
Uniqueid: 1108739972.4

Event: Newexten
Channel: SIP/route886x-5550
Context: sip
Extension: 18
Priority: 2
Application: Wait
AppData: 1
Uniqueid: 1108739972.4

Response: Success
Message: Dual Redirect successful

Event: Unlink
Channel1: SIP/210.201.75.100-08168dd0
Channel2: AsyncGoto/SIP/route886x-5550<ZOMBIE>
Uniqueid1: 1108739916.2
Uniqueid2: 1108739925.3

Event: Hangup
Channel: AsyncGoto/SIP/route886x-5550<ZOMBIE>
Uniqueid: 1108739925.3
Cause: 16






We use this in the astGUIclient to transfer an active conversation(both
parties) to a meetme room:

Action: Redirect
Channel: Zap/73-1
ExtraChannel: SIP/199testphone-1f3c
Exten: 8600029
Context: default
Priority: 1


where 8600029 is a meetme room.

Works very well.

Sadly like most obscure features in Asterisk it is not documented anywhere
very well. But ExtraChannel in Redirect is the only way to send both parties
on a 2-party call into a meetme room so that they can be joined by a 3rd
party(without having a multi-line phone that is).

MATT---


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