[asterisk-users] SIP message routed back to mysql

[mailinglists]

Hello!

* Version: 1.6.0.3-rc1

Scenario: * -> Proxy -> routed back to myself (The only thing changing
is the Request URI)
(And the Record-Route, Via that are added, of course).

Outgoing Context is faxserver-out, incoming context is faxserver (at
least should be).
Outgoing context is straight forward:
[faxserver-out]
exten => _X.,1,NoOP(FAXOUT -- Connecting ${CALLERID(all)} -> ${EXTEN})
exten => _X.,n,SIPAddHeader(P-Preferred-Identity:
<sip:${CALLERID(num)}@test.domain.tld>)
exten => _X.,n,Dial(SIP/${EXTEN}@faxclient,20)
exten => _X.,n,NoOP(FAXOUT -- ${DIALSTATUS} ${CALLERID(num)} ->
${EXTEN})
exten => h,1,NoOP(FAXOUT -- Hangup ${DIALSTATUS} ${CALLERID(num)})

Anyone can give me a direction where to look, since it seems, the *
doesn't even get back to the routing.
Problem is, that the * is routing (?) the incoming SIP request back to
something really strange...

In sip.conf the domain @fax-test.other.domain.tld is bound to context
[faxserver] and a call from somewhere else is terminated there
perfectly.

Do I really have to split outgoing and incoming faxserver onto two *
servers?

br Walter

Log Output:
    -- Executing [055555500011 at faxserver-out:3]
Dial("IAX2/iaxmodem03-4136", "SIP/055555500011 at faxclient,20") in new
stack
Audio is at a.b.c.151 port 17480
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to a.b.c.131:5060:
INVITE sip:055555500011 at test.domain.tld SIP/2.0
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK03b59a6e;rport
Max-Forwards: 70
From: "+43555666" <sip:05555550002 at test.domain.tld>;tag=as3077f211
To: <sip:055555500011 at test.domain.tld>
Contact: <sip:05555550002 at a.b.c.151>
Call-ID: 4218d05b096c3b45278f73e8146561f7 at test.domain.tld
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 291

[...] Authentication skiped
---
    -- Called 055555500011 at faxclient
ast0-1*CLI>
<--- SIP read from UDP://a.b.c.131:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK03b59a6e;rport=5060
From: "+43555666" <sip:05555550002 at test.domain.tld>;tag=as3077f211
To: <sip:055555500011 at test.domain.tld>
Call-ID: 4218d05b096c3b45278f73e8146561f7 at test.domain.tld
CSeq: 102 INVITE
Content-Length: 0

#####################################
Here the call comes back in again, with a different Request URI:
#####################################

<------------->
--- (8 headers 0 lines) ---
ast0-1*CLI>
<--- SIP read from UDP://a.b.c.130:5084 --->
INVITE sip:435555550001 at fax-test.other.domain.tld SIP/2.0
Via: SIP/2.0/UDP a.b.c.130:5084;branch=z9hG4bKMKzUwpTa.L50UQr;rport
Via: SIP/2.0/UDP a.b.c.131;branch=z9hG4bK3fe8.8d63c337.0
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK796c2124;rport=5060
From: "+435555550002"
<sip:05555550002~+43555666 at test.domain.tld>;tag=as3077f211
To: <sip:055555500011 at test.domain.tld>
Call-ID: 4218d05b096c3b45278f73e8146561f7 at test.domain.tld
CSeq: 103 INVITE
Max-Forwards: 68
Supported: replaces,timer,histinfo
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:05555550002 at a.b.c.151>
Content-Length: 291
Content-Type: application/sdp
Record-Route: <sip:a.b.c.130:5084;lr>
Record-Route:
<sip:a.b.c.131;lr;ftag=as3077f211;vsf=AAAAAAAAAAAAAAAAAAAAPl9RQEUdXlRfRV
VcbhURc3QubmVvdGVsLmF0;x-nt-gid=PG00>
User-Agent: NeoTel Media Server~a.b.c.151~a.b.c.151
P-Asserted-Identity: <sip:+435555550002 at test.domain.tld>
Privacy: none
History-Info: <sip:055555500011 at test.domain.tld>;index=1
History-Info:
<sip:+435555550001 at test.domain.tld?Reason=SIP%3Bcause%3D302%3Btext%3D%22
CFU%22>;index=1.1
History-Info: <sip:fax_435555550001 at test.domain.tld>;index=1.2
Date: Thu, 08 Jan 2009 14:20:49 GMT
P-Preferred-Identity: <sip:iiap+43555666 at test.domain.tld>
Remote-Party-ID: <sip:info at os.3play.at>;f=2

#####################################

Here I haven't deleted anything in the log output! 
debug = 9
verbose = 3
sip debug on
If I call from any other source, it is routed to [faxserver,_bla,1]

But instead, the * send's out the call again to 
	INVITE sip:055555500011 at 435555550001 SIP/2.0

Any idea, why?

#####################################


<------------->
--- (25 headers 13 lines) ---
[Jan  8 15:20:49] WARNING[23870]: chan_sip.c:4191 create_addr: No such
host: 435555550001
Audio is at a.b.c.151 port 17480
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to a.b.c.130:5084:
INVITE sip:055555500011 at 435555550001 SIP/2.0
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK7c0c2db0;rport