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-boun...@lists.sip-router.org> on behalf of Ryan Holbein <rtholb...@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 <govoi...@gmail.com<mailto:govoi...@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 <rtholb...@hotmail.com<mailto:rtholb...@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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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