Anca, This was I while back but I finally have time to look into it more. I am on a later version of OpenSIPS and when I use the B2B module I am still getting the "404 not found". Wasn't sure if anyone had anymore ideas on what the issue might be.
On Fri, Dec 3, 2010 at 6:08 AM, Anca Vamanu <[email protected]> wrote: > 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:[email protected]:2074 SIP/2.0 >> Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0 >> To: sip:[email protected]:2074 >> From:<sip:[email protected] <sip%[email protected]> >> >;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:[email protected]: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 <tel:+12312200118>23/12/2001 >> 18<tel:+12312200118>: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:[email protected] <sip%[email protected]> >> >;tag=0f9b47ee30dc18afc732e12a2919b872-aa30 >> To:<sip:[email protected]: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:[email protected]<sip%[email protected]> >> >\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:[email protected]:1850 SIP/2.0 >> Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0 >> From:<sip:[email protected] <sip%[email protected]> >> >;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:[email protected]:5060> >> To:<sip:[email protected] <sip%[email protected]>> >> >> 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 <tel:+12312200118>23/12/2001 >> 18<tel:+12312200118>: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:[email protected] <sip%[email protected]> >> >;tag=0f9b47ee30dc18afc732e12a2919b872-aa30 >> To:<sip:[email protected] <sip%[email protected]>> >> 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:[email protected]: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:[email protected]<sip%[email protected]> >> >;tag=94usbbkjqi >> To:<sip:[email protected] <sip%[email protected]>> >> Call-ID: 3c268edc0da6-3ut9py151hv1 >> CSeq: 2 INVITE >> Max-Forwards: 69 >> Contact:<sip:[email protected]: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..... >> >> > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
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