Yes, I have send you email in detail about my scenario, but Yes, you are right, UA will register to opensips and dial 00123456789, opensips will route that call to Freeswitch, Freeswitch has dialplan to send that call back to opensips removing 00 prefix and then opensips will forward that call to PSTN or outside gateway, Reason i need this solution so later i can add more and more FS without edit any PSTN setting, or my outgoing single IP will be white list.
I have solved issue related 482, and my call routing to outside so that part is working, but again if PSTN callee hangup phone then I am getting error "404 not here" so in short it is not handling BYE properly... I have tried match_dialog() in loose_route() function but didn't help. Here is my config, could you please take a look and let me know if anything missing http://pastebin.com/KL4ZWnM8 also do i need to use record_route() function? On Tue, Sep 2, 2014 at 4:41 AM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Hi Satish, > > SO you have the same call going twice through same FS box ? and when > hitting for the second time you get the 482 ? (once again, for the same > call) > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 01.09.2014 21:34, Satish Patel wrote: > > I have following setup, UA register to Opensips and opensips send call > to FS (freeswitch) and again freeswitch send call back to opensips and then > call get outside routed. in following Senior freeswitch sending 482 Loop > detect error, How do i achieve following scenario? > > [UA]------[Opensips]------[SIP Provider] > | > | > | > [FS] > > > _______________________________________________ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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