[OpenSIPS-Users] B2B issues with To Header (I think)
osiris123d
duane.larson at gmail.com
Thu Nov 11 21:40:47 CET 2010
I am playing with the B2B module and not having a lot of luck. I am using my
original script and adding in the b2b_init_request. I execute all of my
logic like lookup("location") so that the callee info can be set up
correctly. After all of that I do the following
if(is_method("INVITE") && !has_totag()) {
b2b_init_request("refer");
exit;
}
This sends the following request to the callee phone
INVITE sip:9012732009 at 75.XXX.XXX.158:2074 SIP/2.0
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0
To: sip:9012732009 at 75.XXX.XXX.158:2074
From: <sip:9012211612 at irock.com>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
CSeq: 3 INVITE
Call-ID: B2B.114.3927076
Content-Length: 451
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Type: application/sdp
Supported: timer, 100rel, replaces, from-change
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Session-Expires: 3600;refresher=uas
Min-SE: 90
Contact: <sip:b2bua at 173.XXX.XXX.134:5060>
v=0
o=root 535295098 535295098 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 65214 RTP/AVP 9 8 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:et2a2zK91Vh8Hk1o415DWp/kM1BtwbOTmJONkV9E
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--------------------------------------------------------------------------------
Sent to udp:173.XXX.XXX.134:5060 at 23/12/2001 18:15:15:695 (482 bytes):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0
From: <sip:9012211612 at irock.com>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
To: <sip:9012732009 at 75.XXX.XXX.158:2074>
Call-ID: B2B.114.3927076
CSeq: 3 INVITE
User-Agent: snom360/8.4.18
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
Because the TO header doesn't have the real domain on it the phone rejects
it
So I thought by using OpenSIPS local_route I could do the following
local_route {
if (is_method("INVITE")) {
remove_hf("To");
append_hf("To: <sip:9012732004 at coolbeans.com>\r\n");
}
}
This doesn't seem to make a difference at all. The callee phone still
rejects this. here is what the phone does when I use local_route
INVITE sip:9012732004 at 75.XXX.XXX.158:1850 SIP/2.0
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0
From: <sip:9012211612 at irock.com>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
CSeq: 3 INVITE
Call-ID: B2B.464.6147243
Content-Length: 451
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))
Content-Type: application/sdp
Supported: timer, 100rel, replaces, from-change
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Session-Expires: 3600;refresher=uas
Min-SE: 90
Contact: <sip:b2bua at 173.XXX.XXX.134:5060>
To: <sip:9012732004 at coolbeans.com>
v=0
o=root 808120215 808120215 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 64810 RTP/AVP 9 8 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:DXf894oyUu9RbqKk5DGs0bJtaJMlb9zi09qM4S7a
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
--------------------------------------------------------------------------------
Sent to udp:173.XXX.XXX.134:5060 at 23/12/2001 18:05:14:063 (480 bytes):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0
From: <sip:9012211612 at irock.com>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
To: <sip:9012732004 at coolbeans.com>
Call-ID: B2B.464.6147243
CSeq: 3 INVITE
User-Agent: snom870/8.4.18
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Content-Length: 0
Just to be sure I looked an Invite for a call that is good and successful.
INVITE sip:9012732004 at 75.XXX.XXX.158:3072;line=hbpetirz SIP/2.0
Record-Route:
<sip:173.XXX.XXX.134;lr=on;ftag=94usbbkjqi;nat=yes;vst=AAAAAAAAAAAAAAAAAAAACh0ADwlLAgEeFRYcCHI9cGhvbmU-;did=c9b.ac2702a2>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK0dbb.5dfc74b4.0
Via: SIP/2.0/UDP
192.168.33.23:2048;received=75.XXX.XXX.158;branch=z9hG4bK-97gss0xcllrx;rport=2048
From: "Moo 221-1612" <sip:9012211612 at irock.com>;tag=94usbbkjqi
To: <sip:9012732004 at coolbeans.com>
Call-ID: 3c268edc0da6-3ut9py151hv1
CSeq: 2 INVITE
Max-Forwards: 69
Contact: <sip:9012211612 at 75.XXX.XXX.158:2048>;reg-id=1
X-Serialnumber: 0004132902C9
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 453
P-hint: route(3)|setflag7,forcerport,fix_contact
P-hint: inbound->inbound
v=0
o=root 1995837061 1995837061 IN IP4 192.168.33.23
s=call
c=IN IP4 192.168.33.23
t=0 0
m=audio 54868 RTP/AVP 9 8 99 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:+0pSytm8OGoCffuw2hZBe7vu3xGGiRQQafqdOGHA
a=rtpmap:9 g722/8000
a=rtpmap:8 pcma/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
I have no clue why it doesn't work with the local_route edit.....
--
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