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<DIV><FONT size=2>Hi,</FONT></DIV>
<DIV><FONT size=2>We try to do something like somone did in
redirect API, but not fully success...</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>This is what we did, Both channel has been setup and
talking...</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>Action: Redirect<BR>Channel:
SIP/210.201.75.100-081b9170<BR>ExtraChannel:
SIP/route886x-79cb<BR>Exten:18<BR>Context:sip<BR>Priority:1</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>I have two issue:</FONT></DIV>
<DIV><FONT size=2>1. Channel and Extrachannel could be the same tech channel,
sip?</FONT></DIV>
<DIV><FONT size=2>2. Always one certain party connected, one disconnect
//Zombie ?? why??</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV><FONT size=2>
<DIV><BR>Event: Link<BR>Channel1: SIP/210.201.75.100-08168dd0<BR>Channel2:
SIP/route886x-5550<BR>Uniqueid1: 1108739916.2<BR>Uniqueid2: 1108739925.3</DIV>
<DIV> </DIV>
<DIV>Action: Redirect<BR>Channel: SIP/210.201.75.100-08168dd0<BR>ExtraChannel:
SIP/route886x-5550<BR>Exten: 18<BR>Context: sip<BR>Priority: 1</DIV>
<DIV> </DIV>
<DIV>Event: Newchannel<BR>Channel: AsyncGoto/SIP/route886x-5550<BR>State:
Up<BR>Callerid: <unknown><BR>Uniqueid: 1108739972.4</DIV>
<DIV> </DIV>
<DIV>Event: Rename<BR>Oldname: SIP/route886x-5550<BR>Newname:
SIP/route886x-5550<MASQ><BR>Uniqueid: 1108739925.3</DIV>
<DIV> </DIV>
<DIV>Event: Rename<BR>Oldname: AsyncGoto/SIP/route886x-5550<BR>Newname:
SIP/route886x-5550<BR>Uniqueid: 1108739972.4</DIV>
<DIV> </DIV>
<DIV>Event: Rename<BR>Oldname: SIP/route886x-5550<MASQ><BR>Newname:
AsyncGoto/SIP/route886x-5550<ZOMBIE><BR>Uniqueid: 1108739925.3</DIV>
<DIV> </DIV>
<DIV>Event: Newexten<BR>Channel: SIP/route886x-5550<BR>Context:
sip<BR>Extension: 18<BR>Priority: 1<BR>Application:
Answer<BR>AppData:<BR>Uniqueid: 1108739972.4</DIV>
<DIV> </DIV>
<DIV>Event: Newexten<BR>Channel: SIP/route886x-5550<BR>Context:
sip<BR>Extension: 18<BR>Priority: 2<BR>Application: Wait<BR>AppData:
1<BR>Uniqueid: 1108739972.4</DIV>
<DIV> </DIV>
<DIV>Response: Success<BR>Message: Dual Redirect successful</DIV>
<DIV> </DIV>
<DIV>Event: Unlink<BR>Channel1: SIP/210.201.75.100-08168dd0<BR>Channel2:
AsyncGoto/SIP/route886x-5550<ZOMBIE><BR>Uniqueid1:
1108739916.2<BR>Uniqueid2: 1108739925.3</DIV>
<DIV> </DIV>
<DIV>Event: Hangup<BR>Channel:
AsyncGoto/SIP/route886x-5550<ZOMBIE><BR>Uniqueid: 1108739925.3<BR>Cause:
16</DIV>
<DIV> </DIV>
<DIV></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV>We use this in the astGUIclient to transfer an active
conversation(both<BR>parties) to a meetme room:<BR><BR>Action:
Redirect<BR>Channel: Zap/73-1<BR>ExtraChannel: SIP/199testphone-1f3c<BR>Exten:
8600029<BR>Context: default<BR>Priority: 1<BR><BR><BR>where 8600029 is a meetme
room.<BR><BR>Works very well.<BR><BR>Sadly like most obscure features in
Asterisk it is not documented anywhere<BR>very well. But ExtraChannel in
Redirect is the only way to send both parties<BR>on a 2-party call into a meetme
room so that they can be joined by a 3rd<BR>party(without having a multi-line
phone that is).<BR><BR>MATT---<BR><BR><BR></DIV></BODY></HTML>