[asterisk-dev] chan_sip.c / sip.conf : need similar option as "promiscredir" but for REFER and not redirect (302)

tloginbr-asteriskdev at yahoo.com.br tloginbr-asteriskdev at yahoo.com.br
Mon Apr 14 10:03:29 CDT 2008


Hi guys, sorry if this is the wrong list, but I've tried the users
list and forums already, so maybe I'll need to change some code to
get what I need. I need to enable my asterisk (1.4.18.1 running on
debian etch) to accept REFER commands to non local sip address. I've
tried all sorts of configurations inside sip.conf but it always tries
sending my call to a local extension. The scenario is liker this: I
get a call from PSTN and dial a sip address (not in the same machine
as asterisk) that has one sip application (not asterisk). This
application will transfer the call to a second sip address (not
asterisk either) depending on user input. So I need asterisk to
receive the REFER command from my first application and blind
transfer the call to my second application. The problem is that it
always tries sending the call to a local extension. Bellow is a sip
debug from the sip messages. My first application is running on
"sip:5080 at 201.73.67.7:5080" and my second application is running in
"sip:5070 at 201.73.67.7:5070". There many different possible addresses
like this second application and they change, as I load more
application when its needed, so its not really an option to create
extensions for each of this addresses and register them in the
asterisk box. In different part of this system I use redirects (302)
ans the option "promiscredir=yes" inside sip.conf did the job for me,
but in this case I need something to work with REFER.

Thanks for all help,

Thiago


<-- 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






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