31 Aug
2015
31 Aug
'15
4:57 p.m.
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(a)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(a)gmail.com>om>:
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(a)gmail.com>om>:
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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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?
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--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/