<div dir="ltr"><div><div>As i receive the 200 ok without the Record-Route by the gateway, is it possible to the gateway stablish the signalling directly with the user and in this case i didnt receive the bye ?<br><br></div>
i think that is what happen with this gateway<br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014/1/5 Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>Hello Razvan (and everybody), <br><br></div>I try this, the dialog seems to be ok because the dialog is beeing deleted, but i got this messages on syslog<br>
<br>Jan 5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: ERROR:rr:get_remote_target: Invalid routing type - 0<br>
Jan 5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg<br>Jan 5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: In-Dialog BYE from XX.XX.XX.XX (callid=1b1c3705-f0ff-1231-c396-001a4bd5a0b4) is not valid according to dialog<br>
<br><br><br></div><div>in this case, the user send the bye<br><br>U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069<br>BYE sip:255755813256@__IP__OPENSIPS__:5069;transport=udp SIP/2.0.<br>Via: SIP/2.0/UDP __IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport.<br>
Max-Forwards: 70.<br>Contact: <sip:200214@__IP__CUSTOMER__:30664>.<br>To: <sip:255755813256@__IP__OPENSIPS__:5069>;tag=ZSag78XKZQ6yD.<br>From: <<a href="http://sip:200214@209.160.78.42:5069" target="_blank">sip:200214@209.160.78.42:5069</a>>;tag=ea263e5b.<br>
Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA..<br>CSeq: 3 BYE.<br>Proxy-Authorization: Digest username="200214",realm="__IP__OPENSIPS__",nonce="8ef536ce-d39b-4036-90d1-04c54fe9e133",uri="sip:255755813256@__IP__OPENSIPS__:5069;transport=udp",response="72ef0971dda591f6b7684b375a5a3ac1",cnonce="8db354c7dc934c13b3577ff5db18297c",nc=00000002,qop=auth,algorithm=MD5.<br>
User-Agent: Bria Professional release 2.4 stamp 49381.<br>Reason: SIP;description="User Hung Up".<br>Content-Length: 0.<br></div><div><br><br></div><div>And on opensips this is what i have.<br><br>#########<br>
U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079<br>
BYE sip:255755813256@__IP_OPENSIPS:5021;transport=udp SIP/2.0.<br>Via: SIP/2.0/UDP 10.255.2.21:5021;branch=z9hG4bKervg1296366635;received=10.255.2.21.<br>From: <sip:255755813256@__IP_OPENSIPS:5079>;tag=12ab34cd.<br>
To: "200214" <sip:200214@__IP_OPENSIPS>;tag=023883eQv0vHS.<br>Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.<br>CSeq: 23 BYE.<br>Max-Forwards: 70.<br>Content-Length: 0.<br>.<br><br>#<br>U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040<br>
SIP/2.0 481 Call Does Not Exist.<br>Via: SIP/2.0/UDP 10.255.2.21:5021;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__.<br>From: <sip:255755813256@__IP_OPENSIPS:5079>;tag=12ab34cd.<br>To: "200214" <sip:200214@__IP_OPENSIPS>;tag=023883eQv0vHS.<br>
Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.<br>CSeq: 23 BYE.<div class="im"><br>User-Agent: vBilling.<br>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY.<br>Supported: precondition, path, replaces.<br>
</div>
Content-Length: 0.<br><br><br><br></div><div>The call-ID on sip is ok, it is the same of the invite, session progress etc..<br><br></div><div><br></div><div>if you guys have any tip I will apreciate, this is a new situation for me, happen just with this gateway (I dont remember the brand now , it is a friend enviroment and Im trying to help)<br>
</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">2013/12/25 Mike Tesliuk <span dir="ltr"><<a href="mailto:mike@ultra.net.br" target="_blank">mike@ultra.net.br</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div>Hello Razvan, <br><br></div>thank you for your help, i check about this function before, i will try that and i let you know if solve , thank you and happy hollidays<br></div><div><div>
<div class="gmail_extra"><br>
<br><div class="gmail_quote">2013/12/24 Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi, Mike!<br>
<br>
Have you tried matching the dialogs using the match_dialog() function[1]? Also, for sequential requests, you should try using the fix_route_dialog() function[2].<br>
<br>
[1] <a href="http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144" target="_blank">http://www.opensips.org/html/<u></u>docs/modules/1.10.x/dialog.<u></u>html#id295144</a><br>
[2] <a href="http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295287" target="_blank">http://www.opensips.org/html/<u></u>docs/modules/1.10.x/dialog.<u></u>html#id295287</a><br>
<br>
Best regards,<br>
<br>
Răzvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<u></u>com</a><div><br>
<br>
On 12/21/2013 06:25 PM, Mike Tesliuk wrote:<br>
</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>
Hello Guys,<br>
<br>
<br>
Im getting a strange situation here that i dont know how to deal<br>
<br>
i have an enviroment where i have freeswitch receiving a call to billing<br>
and opensips doing the load_balance to the gateways.<br>
<br>
When i send the call to the gateway i receive the reply without the<br>
record-route header, i try to put an asterisk server as gateway and<br>
this not happen in this scenario .<br>
<br>
Below the invite that i send to the gateway<br>
<br></div>
U <a href="http://10.1.69.1:5079" target="_blank">10.1.69.1:5079</a> <<a href="http://10.1.69.1:5079" target="_blank">http://10.1.69.1:5079</a>> -> <a href="http://10.255.2.31:5031" target="_blank">10.255.2.31:5031</a><br>
<<a href="http://10.255.2.31:5031" target="_blank">http://10.255.2.31:5031</a>><br>
INVITE <a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">sip:255755813256@10.1.69.1:<u></u>5079</a><br>
<<a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">http://sip:255755813256@10.1.<u></u>69.1:5079</a>> SIP/2.0.<div><br>
Record-Route: <sip:10.1.69.1:5079;lr;ftag=<u></u>HgcSt10Xa854e;did=9d2.<u></u>723c6252>.<br>
Via: SIP/2.0/UDP 10.1.69.1:5079;branch=<u></u>z9hG4bKe98.72455346.0.<br>
Via: SIP/2.0/UDP<br>
10.1.69.1:5069;received=10.1.<u></u>69.1;rport=5069;branch=<u></u>z9hG4bKK5N8yU10cgage.<br>
Max-Forwards: 68.<br>
From: "200214" <<a href="mailto:sip%3A200214@10.1.69.1" target="_blank">sip:200214@10.1.69.1</a><br></div>
<mailto:<a href="mailto:sip%253A200214@10.1.69.1" target="_blank">sip%3A200214@10.1.69.1</a><u></u>>>;tag=HgcSt10Xa854e.<br>
To: <<a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">sip:255755813256@10.1.69.1:<u></u>5079</a><br>
<<a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">http://sip:255755813256@10.1.<u></u>69.1:5079</a>>>.<div><br>
Call-ID: 4c6591da-e483-1231-6cb4-<u></u>001a4bd5a0b4.<br>
CSeq: 53458861 INVITE.<br>
Contact: <<a href="http://sip:gw+os@10.1.69.1:5069" target="_blank">sip:gw+os@10.1.69.1:5069</a>;<u></u>transport=udp;gw=os>.<br>
User-Agent: vBilling.<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,<br>
REGISTER, NOTIFY.<br>
Supported: precondition, path, replaces.<br>
Allow-Events: talk, hold, conference, refer.<br>
Content-Type: application/sdp.<br>
Content-Disposition: session.<br>
Content-Length: 195.<br>
X-FS-Support: update_display,send_info.<br>
Remote-Party-ID: "200214" <<a href="mailto:sip%3A200214@10.1.69.1" target="_blank">sip:200214@10.1.69.1</a><br></div>
<mailto:<a href="mailto:sip%253A200214@10.1.69.1" target="_blank">sip%3A200214@10.1.69.1</a><u></u>>>;party=calling;screen=yes;<u></u>privacy=off.<div><br>
<br>
<br>
and below the 200 ok that i receive<br>
<br></div>
U <a href="http://10.255.2.31:5031" target="_blank">10.255.2.31:5031</a> <<a href="http://10.255.2.31:5031" target="_blank">http://10.255.2.31:5031</a>> -> <a href="http://10.1.69.1:5079" target="_blank">10.1.69.1:5079</a><br>
<<a href="http://10.1.69.1:5079" target="_blank">http://10.1.69.1:5079</a>><div><br>
SIP/2.0 200 OK.<br>
Via: SIP/2.0/UDP<br>
10.1.69.1:5079;branch=<u></u>z9hG4bKe98.72455346.0;<u></u>received=10.1.69.1.<br>
Via: SIP/2.0/UDP<br>
10.1.69.1:5069;received=10.1.<u></u>69.1;rport=5069;branch=<u></u>z9hG4bKK5N8yU10cgage.<br>
To: <<a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">sip:255755813256@10.1.69.1:<u></u>5079</a><br></div>
<<a href="http://sip:255755813256@10.1.69.1:5079" target="_blank">http://sip:255755813256@10.1.<u></u>69.1:5079</a>>>;tag=12ab34cd.<br>
From: "200214" <<a href="mailto:sip%3A200214@10.1.69.1" target="_blank">sip:200214@10.1.69.1</a><br>
<mailto:<a href="mailto:sip%253A200214@10.1.69.1" target="_blank">sip%3A200214@10.1.69.1</a><u></u>>>;tag=HgcSt10Xa854e.<div><br>
CSeq: 53458861 INVITE.<br>
Call-ID: 4c6591da-e483-1231-6cb4-<u></u>001a4bd5a0b4.<br>
Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.<br>
Supported:.<br>
Allow-Events: telephone-event.<br>
Contact: <<a href="mailto:sip%3A255755813256@10.1.69.1" target="_blank">sip:255755813256@10.1.69.1</a>:<u></u>5031;transport=udp>.<br>
Content-Type: application/sdp.<br>
Content-Length: 196.<br>
<br>
when i send the call to this gateway the loose route did not execute,<br>
and i get error's on dialog because the dialog is not matched<br>
<br>
<br>
how should i deal with a situation like this ?<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br></div>
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</blockquote>
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</blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>