SIP call transfter operation now working

I did get to test sip transfer call transfer, and did get it to work 
with a few simple changes, which will be part of the next release (.10).

If you do a transfer to an "extension", it transfers through the 
originating proxy, unless overriden in the config.  You can also 
transfer to an arbitrary sip uri.

When transfer happens, bayonne drops out of the call (hangs up the 
channel), so the bayonne timeslot is free.  This is commonly called a 
blind transfer.

I have only tested "success" modes so far, so I need to test and 
possibly fix various possible "failure" modes prior to release.

Alexandre BOULANGER wrote:
> Hi,
> 
> Maybe you can give a try to the 'transfer' command
> 
> from scripting doc:
> http://prdownloads.sourceforge.net/gnutelephony/script.pdf?download
> 
> "transfer <number>
> Transfer the call to another location, and then hangup."
> 
> 
> I've never tried this, but would be interested in any feedback if I
> could easily use such a feature in my scripts
> 
> 
> 
> Le lundi 23 janvier 2006 à 13:13 -0300, [email protected] a écrit :
>> Dear friends. Does anybody have a sample script that lets me transfer a
>> call? I have a script where the users can query a database, listen to
>> messages, etc, but I want the same user to be able to be transferred to
>> another number from this script. How can I do it?
>>
>> Thank you all.
>>
>> Roberto From Mendoza, Argentina
>>
>>
>> _______________________________________________
>> Bayonne-devel mailing list
>> [email protected]
>> http://lists.gnu.org/mailman/listinfo/bayonne-devel
>

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