Thanks Sammy, I think I'll stop using SIP realtime. Regards, Ricardo.
On Mon, May 21, 2012 at 5:14 AM, SamyGo <govoi...@gmail.com> wrote: > Hello Ricardo, > The reason why your asterisk refused the calls from phone registering on > SIP proxy is that it only gets INVITE of the call from: a user that is > defined BUT Not Registered within asterisk. > The easy way of solving this is > 1- Stop asterisk SIP realtime and let only the SIP proxy handle > registrations. > 2- Tell asterisk to accept calls from the SIP proxy only (create a SIP > peer for proxy) > This will make everything work. > > Regards, > Sammy. > > On Sat, May 19, 2012 at 9:15 PM, Ricardo Carvalho < > rjcarvalho.li...@gmail.com> wrote: > >> I use an SBC to protect my pool of asterisk servers and as trunking >> endpoint with SIP Telcos. Now I'm trying to implement SIP phone >> registration to be delegated through the SBC, as a proxy. >> >> It doesn't work. It just works when I don't use realtime peers at the >> asterisk servers. Using realtime SIP peers, since there is one SIP phone >> that gets his registration delegated through the SBC, any inbound call that >> tries to reach any asterisk server, coming from any SIP Telco trunk ended >> at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC >> as the IP of the phone that has been registered, it "thinks" that those >> calls coming from the SBC are calls coming from that phone, and it refuses >> them with "401 Unauthorized" replies. I'm using asterisk 1.8.11. >> >> How can I surpass this problem? Is there any configuration that I'm >> lacking on, or is this a limitation of asterisk? >> >> Thanks, >> Ricardo. >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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