[Asterisk-Dev] Manager Redirect - Possible bug ?

[Umar Sear]

Hi there, 

I have being trying to use the Manager API redirect command. 

I have done so with initial sucess, however if I try and redirect an
already redirected channel the far end (calling party) is never sent
an update invite with updated SDP and as a consequence does not hear
anything.

I am using purely SIP and here is a bit more detail.

1. 123456  Calls and is put into a queue
2. I use the manager interface to redirect the call to a SIP phone, sayy 8339
3. Call gets redirected. If 8339 answers, all is well so far. 
4. If I now use the manager API to put the caller on hold (transfer to
an extension 876), the calling party heres no MOH

Redirecting again to 8339 works fine. Looking at ethereal traces it
appears that after step 4 although the cli shows that the call has
been successfully redirected to extension 876 and HOH is started, the
calling party (123456) is still sending RTP to 8339, this is because
it never received intstructions to stop sending.

Is this a bug, or am I missing something. 

Relevant parts of my extensions.conf are reproduced below.

exten => 8339,1,Dial(SIP/8339)

exten => 876,1,Answer
exten => 876,2,MusicOnHold(Default)

Note : If Initially redirect to 876 in the first place, that works and
caller hears music on hold.

Please reply with comments, so that if this is a bug I can raise it as such. 

Thanks

Umar