Hi Paul!
That something I also asked several times and no answer yet. In my
experience, loose_route() is true if the request URI contains the "lr"
paramter (message comes from a strict router). Nevertheless, the
loose_route function will remove its own Route: header.
I solve the troubles by allowing t_relay for all messages with to-tag
(in-dialog).
regards,
klaus
Java Rockx wrote:
Hi All.
I just don't see why loose_route() returned FALSE for this INVITE
message. Can anyone help?
This INVITE was sent from my SIP phone to my SER-0.9 proxy. The IPs
look funny because our SER proxy sits behind a Cisco 3600 and the ALG
stuff has rewritten the IPs (ie, the 10.3.0.221 address).
Regards,
Paul
U 2005/03/14 16:01:19.932196 69.200.205.122:5060 -> 10.3.0.221:5060
INVITE sip:9195531888@66.243.109.99:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 69.200.205.122:5060;branch=z9hG4bK2434592534.
Route: <sip:10.3.0.221;ftag=10000000-0-1811969809;lr>.
Route: <sip:216.229.127.60:5060;lr>.
From: 3475626630 <sip:3475626630@66.243.109.99:5060;user=phone>;tag=550070175.
To: <sip:9195531888@64.152.60.6;user=phone>;tag=10000000-0-1811969809.
Call-ID: 348926-3319820023-453493(a)66.243.109.99.
CSeq: 1 INVITE.
Contact: <sip:3475626630@69.200.205.122:5060>.
max-forwards: 70.
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE.
Content-Type: application/sdp.
Content-Length: 171.
.
v=0.
o=- 9528 3044 IN IP4 0.0.0.0.
s=-.
c=IN IP4 0.0.0.0.
t=0 0.
m=audio 13456 RTP/AVP 18 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendonly.
a=ptime:20.
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