Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk?
Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif <fai...@vopium.com> wrote: > Well a quick n easy fix for you is you can configure you call sending peers > to use username & secret in INVITE. As far as I know it possible in almost > all CISCO, Avaya and all other standard Gateway and SBCs which follows full > SIP RFCs. > > > > If you can’t do it then you need to use curl as realtime engine instead of > MySQL. It will call a URL for each SIP request which you can handle with > flexibility in your CGI scripts with apache. But be careful as per my > experience asterisk 1.6 with curl as realtime engine can handle a max of 120 > registration in parallel if registration refresh time is 120 seconds. > > > > Faisal Hanif > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ricardo Carvalho > *Sent:* Wednesday, February 16, 2011 9:41 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] trunk not working if I register a phone at the > same IP as the trunk peer's IP > > > > How should I configure my asterisk server so that I can receive calls from > an unregistered peer from whom I also receive registrations of sip phones? > > > > I'm asking you this, because with my actual configuration, when I register > a contact from that peer's IP, no more inbound calls are accepted from that > peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication > Required", I assume because they don't carry the registered contact > registration!!! > > My SIP contacts have type=friend and all inbound calls not coming from my > registered phones fall in the default context without authentication, so > that someone in the Internet be able to call freely through the Internet > anyone in my server's dial plan. > > > > Some ideas? > > > > Regards, > > Ricardo Carvalho. > > -- > _____________________________________________________________________ > -- 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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