[Asterisk-Users] IAX2 * -> * handoff

[Darryl Ross]

Hey All,

Yeah yeah, bad form to reply to myself, but <mrgoby> on IRC helped me 
out with the answer just as I sent me question. I'm following up for the 
archives.

Looks like there is an option in the iax.conf file called 
notransfer=yes. Seems to do the same thing as canreinvite=no does in 
sip.conf.

Regards
Darryl



Darryl Ross wrote:

> 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
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