2 May
2012
2 May
'12
8:42 a.m.
Hello,
the Route header in initial invite is usually set by phones that have an
outbound proxy setting. Kamailio doesn't add any Route header itself,
unless append_hf()/insert_hf() is used -- record_route() adds
Record-Route headers.
To deal with this case you should do loose_route() only for requests
within dialog (those that have To header tag). For the rest just remove
the Route header. If you look at default config file in v3.2.x, you will
see this kind of processing (just to analyze it, not need to upgrade to
3.2.x).
Cheers,
Daniel
On 5/2/12 12:26 AM, Geoffrey Mina wrote:
Greetings,
I am confused at some functionality I am seeing with Kamailio 1.5.4.
I know this is an old version, but I don't have the time to go through
a lengthy upgrade process right now. The issue I am seeing is that
the server is inserting a Route header with it's own IP address for an
unknown reason. Here is the initial invite (removed SDP for simplicity):
INVITE sip:13@67.207.130.146:5060 <http://sip:13@67.207.130.146:5060>
SIP/2.0
Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport
From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
<mailto:sip%3A9546496707@dev-asterisk.mydomain.com>>;tag=as1cad6370
To: <sip:13@67.207.130.146:5060 <http://sip:13@67.207.130.146:5060>>
Contact: <sip:9546496707@68.64.220.108
<mailto:sip%3A9546496707@68.64.220.108>>
Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com
<mailto:43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com>
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 70
Remote-Party-ID: "WIRELESS CALLER"
<sip:9546496707@dev-asterisk.mydomain.com
<mailto:sip%3A9546496707@dev-asterisk.mydomain.com>>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Route: <sip:13@boulder-voip.mydomain.com
<mailto:sip%3A13@boulder-voip.mydomain.com>>
P-Account-ID: 99990023
P-Proxy-Route: Yes
Content-Type: application/sdp
Content-Length: 240
The basics of what happen next are:
t_check_trans();
record_route();
remove_hf("P-Proxy-Route");
if(loose_route()){
route(3);
}
route[3]{
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
}
}
The INVITE that goes out has the funky Route: header with the Kamailio
IP in there. This is causing problems for some of the upstream proxy
servers (obviously).
INVITE sip:13@boulder-voip.mydomain.com
<mailto:sip%3A13@boulder-voip.mydomain.com> SIP/2.0
Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370>
Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0
Via: SIP/2.0/UDP
68.64.220.108:5060;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060
From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
<mailto:sip%3A9546496707@dev-asterisk.mydomain.com>>;tag=as1cad6370
To: <sip:13@67.207.130.146:5060 <http://sip:13@67.207.130.146:5060>>
Contact: <sip:9546496707@68.64.220.108
<mailto:sip%3A9546496707@68.64.220.108>>
Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com
<mailto:43134ece101abfca6ecab20212295909@dev-asterisk.mydomain.com>
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 69
Remote-Party-ID: "WIRELESS CALLER"
<sip:9546496707@dev-asterisk.mydomain.com
<mailto:sip%3A9546496707@dev-asterisk.mydomain.com>>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Account-ID: 99990023
Content-Type: application/sdp
Content-Length: 240
Route: <sip:13@67.207.130.146:5060 <http://sip:13@67.207.130.146:5060>>
Any idea what may be causing this to happen and how I could prevent
it? I have tried removing the Route header using the
remove_hf("Route") before doing the t_relay, but that doesn't seem to
help.
Thanks,
Geoff
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