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<br>
"Support broken Registrar:"<br><a href="http://wiki.snom.com/Settings/user_descr_contact">http://wiki.snom.com/Settings/user_descr_contact</a><br><br>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.<br>
<br><br><div class="gmail_quote">On Sun, Jan 30, 2011 at 11:10 PM, Duane Larson <span dir="ltr"><<a href="mailto:duane.larson@gmail.com">duane.larson@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Anca,<br>
<br>
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.<div><div></div><div class="h5"><br><br><br><div class="gmail_quote">On Fri, Dec 3, 2010 at 6:08 AM, Anca Vamanu <span dir="ltr"><<a href="mailto:anca@opensips.org" target="_blank">anca@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi,<br>
<br>
Can you please update your code? There have been a lot of changes and fixes lately in b2b.<br>
<br>
Regards,<br><font color="#888888">
<br>
-- <br>
Anca Vamanu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a></font><div><div></div><div><br>
<br>
<br>
<br>
On 11/11/2010 10:40 PM, osiris123d wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
I am playing with the B2B module and not having a lot of luck. I am using my<br>
original script and adding in the b2b_init_request. I execute all of my<br>
logic like lookup("location") so that the callee info can be set up<br>
correctly. After all of that I do the following<br>
<br>
if(is_method("INVITE")&& !has_totag()) {<br>
b2b_init_request("refer");<br>
exit;<br>
}<br>
<br>
This sends the following request to the callee phone<br>
INVITE sip:9012732009@75.XXX.XXX.158:2074 SIP/2.0<br>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0<br>
To: sip:9012732009@75.XXX.XXX.158:2074<br>
From:<<a href="mailto:sip%3A9012211612@irock.com" target="_blank">sip:9012211612@irock.com</a>>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30<br>
CSeq: 3 INVITE<br>
Call-ID: B2B.114.3927076<br>
Content-Length: 451<br>
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))<br>
Content-Type: application/sdp<br>
Supported: timer, 100rel, replaces, from-change<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,<br>
MESSAGE, INFO, UPDATE<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Contact:<sip:b2bua@173.XXX.XXX.134:5060><br>
<br>
v=0<br>
o=root 535295098 535295098 IN IP4 192.168.33.23<br>
s=call<br>
c=IN IP4 192.168.33.23<br>
t=0 0<br>
m=audio 65214 RTP/AVP 9 8 99 3 18 4 101<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_32<br>
inline:et2a2zK91Vh8Hk1o415DWp/kM1BtwbOTmJONkV9E<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
--------------------------------------------------------------------------------<br>
<br>
Sent to udp:173.XXX.XXX.134:5060 at <a href="tel:+12312200118" target="_blank"></a><a href="tel:+12312200118" target="_blank"></a><a href="tel:+12312200118" target="_blank">23/12/2001 18</a>:15:15:695 (482 bytes):<br>
<br>
SIP/2.0 404 Not Found<br>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1e1.db808976.0<br>
From:<<a href="mailto:sip%3A9012211612@irock.com" target="_blank">sip:9012211612@irock.com</a>>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30<br>
To:<sip:9012732009@75.XXX.XXX.158:2074><br>
Call-ID: B2B.114.3927076<br>
CSeq: 3 INVITE<br>
User-Agent: snom360/8.4.18<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,<br>
MESSAGE, INFO, UPDATE<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, 100rel, replaces, from-change<br>
Content-Length: 0<br>
<br>
<br>
<br>
Because the TO header doesn't have the real domain on it the phone rejects<br>
it<br>
<br>
So I thought by using OpenSIPS local_route I could do the following<br>
local_route {<br>
if (is_method("INVITE")) {<br>
remove_hf("To");<br>
append_hf("To:<<a href="mailto:sip%3A9012732004@coolbeans.com" target="_blank">sip:9012732004@coolbeans.com</a>>\r\n");<br>
}<br>
}<br>
<br>
<br>
<br>
This doesn't seem to make a difference at all. The callee phone still<br>
rejects this. here is what the phone does when I use local_route<br>
<br>
<br>
INVITE sip:9012732004@75.XXX.XXX.158:1850 SIP/2.0<br>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0<br>
From:<<a href="mailto:sip%3A9012211612@irock.com" target="_blank">sip:9012211612@irock.com</a>>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30<br>
CSeq: 3 INVITE<br>
Call-ID: B2B.464.6147243<br>
Content-Length: 451<br>
User-Agent: OpenSIPS (1.6.3-notls (x86_64/linux))<br>
Content-Type: application/sdp<br>
Supported: timer, 100rel, replaces, from-change<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,<br>
MESSAGE, INFO, UPDATE<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Contact:<sip:b2bua@173.XXX.XXX.134:5060><br>
To:<<a href="mailto:sip%3A9012732004@coolbeans.com" target="_blank">sip:9012732004@coolbeans.com</a>><br>
<br>
v=0<br>
o=root 808120215 808120215 IN IP4 192.168.33.23<br>
s=call<br>
c=IN IP4 192.168.33.23<br>
t=0 0<br>
m=audio 64810 RTP/AVP 9 8 99 3 18 4 101<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_32<br>
inline:DXf894oyUu9RbqKk5DGs0bJtaJMlb9zi09qM4S7a<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
<br>
--------------------------------------------------------------------------------<br>
<br>
Sent to udp:173.XXX.XXX.134:5060 at <a href="tel:+12312200118" target="_blank"></a><a href="tel:+12312200118" target="_blank"></a><a href="tel:+12312200118" target="_blank">23/12/2001 18</a>:05:14:063 (480 bytes):<br>
<br>
SIP/2.0 404 Not Found<br>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK1a0c.7a9053f6.0<br>
From:<<a href="mailto:sip%3A9012211612@irock.com" target="_blank">sip:9012211612@irock.com</a>>;tag=0f9b47ee30dc18afc732e12a2919b872-aa30<br>
To:<<a href="mailto:sip%3A9012732004@coolbeans.com" target="_blank">sip:9012732004@coolbeans.com</a>><br>
Call-ID: B2B.464.6147243<br>
CSeq: 3 INVITE<br>
User-Agent: snom870/8.4.18<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,<br>
MESSAGE, INFO, UPDATE<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, 100rel, replaces, from-change<br>
Content-Length: 0<br>
<br>
<br>
<br>
<br>
<br>
<br>
Just to be sure I looked an Invite for a call that is good and successful.<br>
<br>
INVITE sip:9012732004@75.XXX.XXX.158:3072;line=hbpetirz SIP/2.0<br>
Record-Route:<br>
<sip:173.XXX.XXX.134;lr=on;ftag=94usbbkjqi;nat=yes;vst=AAAAAAAAAAAAAAAAAAAACh0ADwlLAgEeFRYcCHI9cGhvbmU-;did=c9b.ac2702a2><br>
Via: SIP/2.0/UDP 173.XXX.XXX.134;branch=z9hG4bK0dbb.5dfc74b4.0<br>
Via: SIP/2.0/UDP<br>
192.168.33.23:2048;received=75.XXX.XXX.158;branch=z9hG4bK-97gss0xcllrx;rport=2048<br>
From: "Moo 221-1612"<<a href="mailto:sip%3A9012211612@irock.com" target="_blank">sip:9012211612@irock.com</a>>;tag=94usbbkjqi<br>
To:<<a href="mailto:sip%3A9012732004@coolbeans.com" target="_blank">sip:9012732004@coolbeans.com</a>><br>
Call-ID: 3c268edc0da6-3ut9py151hv1<br>
CSeq: 2 INVITE<br>
Max-Forwards: 69<br>
Contact:<sip:9012211612@75.XXX.XXX.158:2048>;reg-id=1<br>
X-Serialnumber: 0004132902C9<br>
P-Key-Flags: resolution="31x13", keys="4"<br>
User-Agent: snom360/8.4.18<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,<br>
MESSAGE, INFO, UPDATE<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, 100rel, replaces, from-change<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Content-Type: application/sdp<br>
Content-Length: 453<br>
P-hint: route(3)|setflag7,forcerport,fix_contact<br>
P-hint: inbound->inbound<br>
<br>
v=0<br>
o=root 1995837061 1995837061 IN IP4 192.168.33.23<br>
s=call<br>
c=IN IP4 192.168.33.23<br>
t=0 0<br>
m=audio 54868 RTP/AVP 9 8 99 3 18 4 101<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_32<br>
inline:+0pSytm8OGoCffuw2hZBe7vu3xGGiRQQafqdOGHA<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
<br>
I have no clue why it doesn't work with the local_route edit.....<br>
<br>
</blockquote>
</div></div></blockquote></div><br><br clear="all"><br></div></div>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br><font color="#888888">--<br>
</font></blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>