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[tloginbr-asterisk at y...]

Hi everyone,

I'm currently trying to enable call transfer to different domains in
asterisk box (Asterisk 1.2.13 running on Debian etch). I have a
configuration that requires me to transfer call to separate domains
like ext at 10.10.10.10:5050. My calls come from a R2 channels in a
board installed in the machine. When the call comes in I dial a sip
address in another machine and I need to receive REFER from this
other machine to transfer the call to a third sip URI, that may be or
not in any of the two machines . My machines change all the time, so
registering them in my asterisk box is not an option. The big picture
here is this: I have a asterisk box to receive calls from PSTN and I
send this calls to sip application that I made that will transfer the
call from the user to different sip application depending on user
input. And this other application also need the ability to transfer
calls to different sip URI. The applications are conferences, voice
mail and others, each running on a different sip uri (ext at ip:port)
and the user needs to jump between them. So I need my asterisk box to
accept arbitrary sip URI in a REFER (xfer) command. Right now it
always tries to send the call to a local extension, for example, if I
have a call from my asterisk to "555 at 10.10.10.1:5060" and this
application asks asterisk to transfer this call to
"666 at 10.10.10.2:5070" asterisk will try to send the to the local
extension 666. Bellow I have a sip debug from the messages. My
asterisk box is running in the IP 201.73.67.5, and my first
application (the one that asterisk dials directly) is at the address
201.73.67.7:5080 and it transfers the calls to 201.73.67.7:5070, but
it fails.

All help is very much welcome.

Thanks in advance,

Thiago

Sip debug:

<-- 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


---
set_destination: Parsing <sip:201.73.67.7:5080> for address/port to
send to
set_destination: set destination to 201.73.67.7, port 5080
Reliably Transmitting (no NAT) to 201.73.67.7:5080:
NOTIFY sip:201.73.67.7:5080 SIP/2.0
Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK26db8c59;rport
From: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
To: <sip:0778 at 201.73.67.7:5080>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
Contact: <sip:3130296800 at 201.73.67.5>
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=15651
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
set_destination: Parsing <sip:201.73.67.7:5080> for address/port to
send to
set_destination: set destination to 201.73.67.7, port 5080
Reliably Transmitting (no NAT) to 201.73.67.7:5080:
BYE sip:201.73.67.7:5080 SIP/2.0
Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK1e66e326;rport
From: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
To: <sip:0778 at 201.73.67.7:5080>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
Content-Length: 0


---

<-- SIP read from 201.73.67.7:5080:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK26db8c59
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
From: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
To: <sip:0778 at 201.73.67.7>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
CSeq: 103 NOTIFY
Contact: <sip:201.73.67.7:5080>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
PUBLISH, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, norefersub
Content-Length: 0


--- (10 headers 0 lines) ---

<-- SIP read from 201.73.67.7:5080:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
201.73.67.5:5060;rport=5060;received=201.73.67.5;branch=z9hG4bK1e66e326
Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5
From: "3130296800" <sip:3130296800 at 201.73.67.5>;tag=as26b5df58
To: <sip:0778 at 201.73.67.7>;tag=1jAy-XotYlPo06lq7VDTkQxfne5PnnPA
CSeq: 104 BYE
Content-Length: 0


--- (7 headers 0 lines) ---


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