Hi, Mike!
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].
[1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144
[2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295287
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
Hello Guys,
Im getting a strange situation here that i dont know how to deal
i have an enviroment where i have freeswitch receiving a call to billing
and opensips doing the load_balance to the gateways.
When i send the call to the gateway i receive the reply without the
record-route header, i try to put an asterisk server as gateway and
this not happen in this scenario .
Below the invite that i send to the gateway
U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031
<http://10.255.2.31:5031>
INVITE sip:[email protected]:5079
<http://sip:[email protected]:5079> SIP/2.0.
Record-Route: <sip:10.1.69.1:5079;lr;ftag=HgcSt10Xa854e;did=9d2.723c6252>.
Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.
Via: SIP/2.0/UDP
10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
Max-Forwards: 68.
From: "200214" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=HgcSt10Xa854e.
To: <sip:[email protected]:5079
<http://sip:[email protected]:5079>>.
Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
CSeq: 53458861 INVITE.
Contact: <sip:[email protected]:5069;transport=udp;gw=os>.
User-Agent: vBilling.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, NOTIFY.
Supported: precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 195.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "200214" <sip:[email protected]
<mailto:sip%[email protected]>>;party=calling;screen=yes;privacy=off.
and below the 200 ok that i receive
U 10.255.2.31:5031 <http://10.255.2.31:5031> -> 10.1.69.1:5079
<http://10.1.69.1:5079>
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
10.1.69.1:5079;branch=z9hG4bKe98.72455346.0;received=10.1.69.1.
Via: SIP/2.0/UDP
10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
To: <sip:[email protected]:5079
<http://sip:[email protected]:5079>>;tag=12ab34cd.
From: "200214" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=HgcSt10Xa854e.
CSeq: 53458861 INVITE.
Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.
Supported:.
Allow-Events: telephone-event.
Contact: <sip:[email protected]:5031;transport=udp>.
Content-Type: application/sdp.
Content-Length: 196.
when i send the call to this gateway the loose route did not execute,
and i get error's on dialog because the dialog is not matched
how should i deal with a situation like this ?
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