[SR-Users] Using PSTN as "fallback"

[Alexandru Covalschi]

Or well, if you don't use any media-relay you just need
http://kamailio.org/docs/modules/4.4.x/modules/auth_db.html#idp15567480

2015-08-30 22:41 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> it depends on which PBX you use for media relay and which codes when no
> user available does it return.
> What I'd suggest is to check if call is coming not from PSTN (if it comes
> from PSTN - it's for sure must be routed to PBX) and if TRUE, then first
> send call to PBX and if answer is not 180/183 200 etc. (you can catch that
> in a specific failure_route) route calls back to PSTN.
>
> 2015-08-30 12:04 GMT+03:00 Michael Nielsen <mic.niel84 at gmail.com>:
>
>> I have Kamailio running and connected to a PSTN gateway.
>>
>> My subscribers are named ex. +442071234567 - same as their real GSM
>> number from my PSTN gateway.
>>
>> I'm using the standard kamailio.cfg which ships with version 4.3.
>>
>> When I'm trying to dial SIP client to SIP client I would like to have
>> Kamailio route the call internally if a subscriber exists with ex.
>> +442071234567.
>> If no subscriber exists with ex. +442071234567 it should send it to my
>> PSTN gateway.
>>
>> As it is now it seems as if it are trying to both call "internally" and
>> via the PSTN gateway.
>>
>> How should one fix this issue the best way?
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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