[OpenSIPS-Users] Opensips with freeswitch strange 482 error

[Satish Patel]

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 <bogdan at 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 at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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