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.<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">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 class="h5"><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">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">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>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>