[OpenSIPS-Users] B2B issues with To Header (I think)
Anca Vamanu
anca at opensips.org
Fri Dec 3 13:08:53 CET 2010
Hi,
Can you please update your code? There have been a lot of changes and
fixes lately in b2b.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 11/11/2010 10:40 PM, osiris123d wrote:
> 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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