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 ?
i think that is what happen with this gateway 2014/1/5 Mike Tesliuk <[email protected]> > Hello Razvan (and everybody), > > I try this, the dialog seems to be ok because the dialog is beeing > deleted, but i got this messages on syslog > > Jan 5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: > ERROR:rr:get_remote_target: Invalid routing type - 0 > Jan 5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: > ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg > 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 > > > > in this case, the user send the bye > > U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069 > BYE sip:255755813256@__IP__OPENSIPS__:5069;transport=udp SIP/2.0. > Via: SIP/2.0/UDP > __IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport. > Max-Forwards: 70. > Contact: <sip:200214@__IP__CUSTOMER__:30664>. > To: <sip:255755813256@__IP__OPENSIPS__:5069>;tag=ZSag78XKZQ6yD. > From: <sip:[email protected]:5069>;tag=ea263e5b. > Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA.. > CSeq: 3 BYE. > 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. > User-Agent: Bria Professional release 2.4 stamp 49381. > Reason: SIP;description="User Hung Up". > Content-Length: 0. > > > And on opensips this is what i have. > > ######### > U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079 > BYE sip:255755813256@__IP_OPENSIPS:5021;transport=udp SIP/2.0. > Via: SIP/2.0/UDP 10.255.2.21:5021 > ;branch=z9hG4bKervg1296366635;received=10.255.2.21. > From: <sip:255755813256@__IP_OPENSIPS:5079>;tag=12ab34cd. > To: "200214" <sip:200214@__IP_OPENSIPS>;tag=023883eQv0vHS. > Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4. > CSeq: 23 BYE. > Max-Forwards: 70. > Content-Length: 0. > . > > # > U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040 > SIP/2.0 481 Call Does Not Exist. > Via: SIP/2.0/UDP 10.255.2.21:5021 > ;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__. > From: <sip:255755813256@__IP_OPENSIPS:5079>;tag=12ab34cd. > To: "200214" <sip:200214@__IP_OPENSIPS>;tag=023883eQv0vHS. > Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4. > CSeq: 23 BYE. > > User-Agent: vBilling. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > NOTIFY. > Supported: precondition, path, replaces. > Content-Length: 0. > > > > The call-ID on sip is ok, it is the same of the invite, session progress > etc.. > > > 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) > > > 2013/12/25 Mike Tesliuk <[email protected]> > >> Hello Razvan, >> >> 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 >> >> >> 2013/12/24 Răzvan Crainea <[email protected]> >> >>> 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 ? >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >
_______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
