[OpenSIPS-Users] B2B issues with To Header (I think)

Duane Larson duane.larson at gmail.com
Mon Jan 31 06:10:44 CET 2011 

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 <anca at opensips.org> 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: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 <sip%3A9012211612 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 <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:9012211612 at irock.com <sip%3A9012211612 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<sip%3A9012732004 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 <sip%3A9012211612 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 <sip%3A9012732004 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 <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:9012211612 at irock.com <sip%3A9012211612 at irock.com>
>> >;tag=0f9b47ee30dc18afc732e12a2919b872-aa30
>> To:<sip:9012732004 at coolbeans.com <sip%3A9012732004 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<sip%3A9012211612 at irock.com>
>> >;tag=94usbbkjqi
>> To:<sip:9012732004 at coolbeans.com <sip%3A9012732004 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.....
>>
>>
>


-- 
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
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