[asterisk-users] Similar option as promiscredir to use in transfer (REFER)

[tloginbr-asteriskusers at yahoo.com.br]

Thanks for the reply, Johansson. Sorry if my question was not very
clear... What I need is that asterisk accepts a REFER command from
the client, sending the call to a non local domain. The scenario is
this: I receive a call from PSTN and dial a sip address that contains
one of my applications (running in a separate machine). This
application receives input from the user and then transfers the call
to another application (in a third machine). The call from PSTN is
going to be in asterisk (that got the call in first place) all the
time, just the other end will change depending on user input. Bellow
is a sip debug from this operation. Asterisk is running in
201.73.67.5:5060 and my first application is at 
"5080 at 201.73.67.7:5080". This application then tries to transfer the
call to a second application located at "5070 at 201.73.67.7:5070", but
asterisk ignores the part after the "@" from the uri and tries
sending the call to the extension 5070 in the context
"from-sip-external". I had a similar problem with redirects (302),
but I solved it using the option promiscredir=yes inside "sip.conf".
I've already tried setting the option "domain=" in sip.conf but that
didn't help...



<-- SIP read from 201.73.67.7:5080:
REFER sip:3130296800 at 201.73.67.5 SIP/2.0
Via: SIP/2.0/UDP
201.73.67.7:5080;rport;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx
Max-Forwards: 70
From: <sip:0778 at 201.73.67.7>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
To: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
Contact: <sip:201.73.67.7:5080>
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 15651 REFER
Event: refer
Expires: 300
Accept: message/sipfrag;version=2.0
Allow-Events: presence, refer
Refer-To: sip:5070 at 201.73.67.7:5070
Referred-By: <sip:0778 at 201.73.67.7>
Content-Length:  0


--- (15 headers 0 lines) ---
Transfer to 5070 in from-sip-external
Transfer from 0778 in from-sip-external
Transmitting (no NAT) to 201.73.67.7:5080:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
201.73.67.7:5080;branch=z9hG4bKPj3r0RqvljQLyTKpBVXgbhce5dADV20tVx;received=201.73.67.7;rport=5080
From: <sip:0778 at 201.73.67.7>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
To: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 15651 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3130296800 at 201.73.67.5>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing



Thiago





> 
> 13 apr 2008 kl. 17.46 skrev <tloginbr-asteriskusers at yahoo.com.br>:
> > I made a similar question in a previous thread, but there was no
> > answer, so I think I was not very clear making the question. What
> I
> > need is some configuration that works like "promiscredir=yes" in
> > sip.conf that enables me to do the same thing with transfer
> (REFER),
> > letting me transfer a sip call to a non local sip address.
> 
> 
> I'm still not really sure what you ask for, but I'll give it a try.
> 
> The transfer() dialplan application supports generating a REFER
> from  
> Asterisk to the client. If the call is not answered, it will send
> 302,  
> if the call is in UP state (answered), Asterisk will send a REFER.
> Try  
> it.
> 
> Best regards,
> /Olle
> 
> 
> ---
> * Olle E. Johansson - oej at edvina.net
> * Asterisk Training http://edvina.net/training/
> * Asterisk SIP Masterclass, Orlando, Florida Next week
> * A few seats left - register today!
> 
> 



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