1.8 supports static peers along with realtime peers. I have tested. On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho < [email protected]> wrote:
> Thanks Faisal, in fact I made a test that confirmed that in realtime > asterisk doesn’t supported static peers, like you told me. > Do you know if newer versions of asterisk, like 1.8, have this issue > already solved? > > Regards, > Ricardo. > > > > > On Wed, Feb 16, 2011 at 6:26 PM, Faisal Hanif <[email protected]> wrote: > >> I have played a lot on this issue with asterisk config but in realtime it >> doesn’t supported static peers with version 1.6.2.14. >> >> >> >> *From:* Ricardo Carvalho [mailto:[email protected]] >> *Sent:* Wednesday, February 16, 2011 10:21 PM >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Cc:* Faisal Hanif >> *Subject:* Re: [asterisk-users] trunk not working if I register a phone >> at the same IP as the trunk peer's IP >> >> >> >> 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 <[email protected]> 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:* [email protected] [mailto: >> [email protected]] *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 >> >> >> > > > -- > _____________________________________________________________________ > -- 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 > -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 3333 6767 26 E: [email protected] W: www.axvoice.com
-- _____________________________________________________________________ -- 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
