9 Nov
2015
9 Nov
'15
6:57 p.m.
Hello,
I was wondering if anyone used Siremis GUI to setup IP based Auth. Also once that is done
how would I route those to my AST machine. Right now In test I have OUTSIDE ->
Kamailio -> AST Box ... I would like to do this though the SIREMIS Webpage so does
anyone have any idea? I would like to try this before I need to install Asterisk just for
that reason.. I am only using Kamailio for routing/Ip based Auth while my AST box handles
all Calls/Voicemail/IVR.
Thanks for the help
________________________________
From: sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of Ryan Holbein
<rtholbein(a)hotmail.com>
Sent: Saturday, November 7, 2015 4:36 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Examples
Thank you Sammy! I will have to give this a try and follow your rules. Looks likes. Great
start!! Is there a way of doing this though the siremis webpage or has to be done via
kamialio config file?
Sent from my iPhone
On Nov 6, 2015, at 5:51 PM, SamyGo
<govoiper@gmail.com<mailto:govoiper@gmail.com>> wrote:
Hi Ryan,
Where are your trunks !?
if your provider can just send calls to your IP address then just do IP based
authentication in Kamailio and once provider is authenticated relay the call to the
Internal PBX.
so with reference to the code here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I will try to
guide you.
1 - allow IP AUTHENTICATION by adding line
#define WITH_IPAUTH
after the line saying "#define WITH_AUTH"
2 - Put the IP address plus port of the provider in "permission" database table
and restart Kamailio (for first time only) for next time you make changes in that table
execute this command
Linux:~#kamctl address reload
3 - Now everytime your provider sends a call it will be accepted BUT the call still needs
to be routed to the internal PBX.
4 - since WITH_ASTERISK is defined on top as well so Kamailio will check the IP address of
your internal PBX from this:
asterisk.bindip = "192.168.178.25" desc "Asterisk IP Address"
asterisk.bindport = "5080" desc "Asterisk Port"
If you want to have a different criteria to route call to internal PBX like Load-Balancing
or decide based on DID the calls goes to a specific server, or based on accound it routes
to a specific PBX then thats your logic and should be handled inside the route[TOASTERISK]
- similarly route[FROMASTERISK] needs changes to allow calls coming back from Internal
PBXs.
I hope it just gives you some idea of what to do next.
Regards,
Sammy
On Fri, Nov 6, 2015 at 12:25 PM, Ryan Holbein
<rtholbein@hotmail.com<mailto:rtholbein@hotmail.com>> wrote:
Hello,
I have everything setup and installed... Does anyone have a good link or could tell me the
steps of how to connect my trunks to phone provider and then another one would be how to
route the calls to the internal PBX system.
Thank you
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