Oh thanks Olle, I'm on the wrong list. Sorry about that.
Från: [email protected] [mailto:[email protected]] För Olle E. Johansson Skickat: den 25 november 2013 11:32 Till: Asterisk Developers Mailing List Kopia: Olle E Johansson Ämne: Re: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ? Hi! The development mailing list is not a good list for asking general questions - please use the user's mailing list for that. There's currently no way of authenticating devices that are not peers or users of asterisk. Registration is unrelated to other requests. You could do this with Kamailio of course. Hälsningar /Olle On 25 Nov 2013, at 11:17, Johan Sandgren <[email protected]<mailto:[email protected]>> wrote: Did I catch you all at a bad time with the code reviews and all? I'm still hoping for some responses or ideas, or from someone with knowledge. Are there anyone out there? :) /Johan Från: [email protected]<mailto:[email protected]> [mailto:[email protected]] För Johan Sandgren Skickat: den 22 november 2013 18:09 Till: [email protected]<mailto:[email protected]> Ämne: [asterisk-dev] Make asterisk V10 require SIP authentication for unregistered Message / Invite (calls) ? Hi everyone, Is it possible to make asterisk REQUIRE authentication for unregistered incoming SIP MESSAGE och SIP INVITE (all related to incoming calls)? With registered sip clients, asterisk successfully asks for authorization for each message. I also need to support unregistered clients (it would be a global user + password in this case). Any ideas of which global settings does this? I haven't found anything yet. Or suggestions of where I possibly could edit the source code to enable this feature. I have a bit of knowledge of the sourcecode, and have compiled it before. Johan Sandgren Software Engineer Svep Design Center AB S:t Lars väg 42A 222 70 Lund, Sweden Phone +46 46 192 722 E-mail [email protected]<mailto:[email protected]> Website www.svep.se<http://www.svep.se/> -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com<http://www.api-digital.com/> -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
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