[asterisk-users] Function transfer RFC 5589

[David Pinedo]

Hello,

I have the following scenario:

   1. VoIP Gateway G400 connected to PSTN
   2. Asterisk server 1 (working as IVR)
   3. Asterisk server 2 (working as ACD, with several agents connected)

I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk
server 1 (IVR). When the IVR ends working with the call, transfers it to
the Asterisk server 2 (ACD).

In Asterisk server 1 (IVR) I'm using the function Transfer()
<http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer>, which sends a
SIP REFER to the VoIP Gateway, in the way that is explained in RFC 5589
http://tools.ietf.org/html/rfc5589#section-6.1
Where:
VoIP Gateway G400 is de Transferee
Asterisk server 1 (IVR) is the Transferor
And Asterisk server 2 (ACD) is the Transfer Target

The SIP transaction is completed correctly, with the difference that there
is no "INVITE (hold)" from Transferor to Transferee.

In the G400 once finalized the transaction there is no audio: I think is a
problem in the G400 because I have done the same test with a Vega gateway
and with a softphone and the call is transferred correctly (also audio).
1) Does any one know if exists any problem with transferences, in this
gateway?

By other side, I'd like to do an "attended transfer" as is explained in the
same RFC 5589
http://tools.ietf.org/html/rfc5589#page-24
2) Is it possible to do that with Transfer function?

3) There is another way (different to use transfer function) to do this
kind of transferences?

Thanks in advance

-- 

*David Pinedo García*
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