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)}[email protected]>)
exten => _X.,n,Dial(SIP/${ext...@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 [055555500...@faxserver-out:3]
Dial("IAX2/iaxmodem03-4136", "SIP/055555500...@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:[email protected] SIP/2.0
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK03b59a6e;rport
Max-Forwards: 70
From: "+43555666" <sip:[email protected]>;tag=as3077f211
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 291
[...] Authentication skiped
---
-- Called 055555500...@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:[email protected]>;tag=as3077f211
To: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected] 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:[email protected]>;tag=as3077f211
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Max-Forwards: 68
Supported: replaces,timer,histinfo
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:[email protected]>
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:[email protected]>
Privacy: none
History-Info: <sip:[email protected]>;index=1
History-Info:
<sip:[email protected]?reason=sip%3bcause%3d302%3btext%3d%22
CFU%22>;index=1.1
History-Info: <sip:[email protected]>;index=1.2
Date: Thu, 08 Jan 2009 14:20:49 GMT
P-Preferred-Identity: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]>;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:055555500...@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:055555500...@435555550001 SIP/2.0
Via: SIP/2.0/UDP a.b.c.151:5060;branch=z9hG4bK7c0c2db0;rport
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