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

Duane Larson duane.larson at gmail.com
Mon Jan 31 06:26:31 CET 2011 

So I sent the last email and then noticed that the good call's INVITE
message had a difference with the contact header and the INVITE header.  So
with the Snom config I looked around and had the option to do what they call
"Support broken Registrar:"
http://wiki.snom.com/Settings/user_descr_contact

When I enable this and make a call to the snom phone it is now ringing
instead of the caller getting the "404 Not Found", but when the callee picks
up the callers phone is still ringing like the callee never picked up.  Is
this something wrong with the B2B module?  I would think other people would
run into this too.


On Sun, Jan 30, 2011 at 11:10 PM, Duane Larson <duane.larson at gmail.com>wrote:

> 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><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><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
> *--*--*--*--*--*
> --
>



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