[Asterisk-Users] IAX2 * -> * handoff

[Darryl Ross]

Hey All,

I am setting up a network of Asterisk servers using IAX2. I am wondering 
if it is possible to disable the handoff feature?

At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is 
centrally hosted in a data centre. In addition the central machine has 
an IAX2 link to a VOIP provider (and might be set up with more in the 
future). All calls are routed through that central machine.

When a call is made at the moment, say from asterisk1 to asterisk2, the 
central machine does a native bridge of the channels (I assume this 
means that no codec translation needs to be done) and then gets both 
ends to transfer to each other. Then it releases the channels. I want to 
stop that occuring. I guess the SIP name for this is ReInviting. The 
logs look like this (passwords changed):

---------------------------------------------------------------------------------------

     -- Accepting AUTHENTICATED call from 203.221.53.223, requested 
format = 2, actual format = 2
     -- Executing Dial("IAX2[darryl at darryl]/6", 
"IAX2/oeg_pbx:password at mike/100,60,tT") in new stack
     -- Called oeg_pbx:password at mike/100
     -- Call accepted by 203.31.11.15 (format GSM)
     -- Format for call is GSM
     -- IAX2[mike]/14 stopped sounds
     -- IAX2[mike]/14 is ringing
     -- IAX2[mike]/14 stopped sounds
     -- IAX2[mike]/14 answered IAX2[darryl at darryl]/6
     -- Attempting native bridge of IAX2[darryl at darryl]/6 and IAX2[mike]/14
     -- Channel 'IAX2[darryl at darryl]/6' ready to transfer
     -- Channel 'IAX2[mike]/14' ready to transfer
     -- Releasing IAX2[mike]/14 and IAX2[darryl at darryl]/6
     -- Hungup 'IAX2[mike]/14'
   == Spawn extension (pstn-authorised, 3444100, 1) exited non-zero on 
'IAX2[darryl at darryl]/6'
     -- Hungup 'IAX2[darryl at darryl]/6'

---------------------------------------------------------------------------------------

The reason this is a problem is that the CDR records on the central 
machine only show a 6 second call or thereabouts, where we need it to 
keep track of the entire call.

I've tried using the 't' and 'T' options for the Dial command, which 
according to the docs is supposed to make it stay in the media path, but 
it still hands the calls off.

Any ideas??

TIA
Darryl