[OpenSIPS-Users] Opensips with freeswitch strange 482 error

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Sep 3 09:10:26 CEST 2014


Hey,

your script does not do any create_dialog() + topology_hiding() for the 
initial INVITEs - so the TH will not work for you, the calls will simply 
work in normal way.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02.09.2014 19:53, Satish Patel wrote:
> 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 <mailto: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 Developer
>     http://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 list
>>     Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>

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