[SR-Users] Using PSTN as "fallback"

Michael Nielsen mic.niel84 at gmail.com
Mon Aug 31 08:57:35 CEST 2015


I'll look at the auth_db, as I'm not routing to any media-relay on simple
calls between subscribers.
Only for voicemail and such do I route the calls to my FreeSWITCH.

On Sun, Aug 30, 2015 at 9:50 PM, Alexandru Covalschi <568691 at gmail.com>
wrote:

> 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/
>
> _______________________________________________
> 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
>
>
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