Please post some configurations. YOU CAN REMOVE username / secret from the sip configs but posting remaining configs will only help to your issue resolution, Alex first words were "This is just a speculative shot in the dark" .
Fine if you were just Trolling here ! 2011/9/30 cnasterisk <[email protected]> > ** > asterisk can register successfully on the remote party. so i think username > & password must be ok > > > 2011-09-30 > ------------------------------ > cnasterisk > ------------------------------ > *>发件人：* Sam Govind > *>发送时间：* 2011-09-30 16:05:21 > *>收件人：* Asterisk Users Mailing List - Non-Commercial Discussion > *>抄送：* > *>主题：* Re: [asterisk-users] invite authentication error !? > >Whatever that remote party is, you are most definitely using > a username/secret declaration for that. So the sip attributes set for that > proxy define the behaviour for this. > > On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk <[email protected]> wrote: > >> ** >> hi, Sam >> thanks for your kindly reply. >> The remote proxy is not asterisk >> >> 2011-09-30 >> ------------------------------ >> cnasterisk >> ------------------------------ >> *>发件人：* Sam Govind >> *>发送时间：* 2011-09-30 15:36:41 >> *>收件人：* Asterisk Users Mailing List - Non-Commercial Discussion >> *>抄送：* >> *>主题：* Re: [asterisk-users] invite authentication error !? >> >What Sip declaration are you using for the remote sip proxy in >> sip.conf? >> >> On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov < >> [email protected]> wrote: >> >>> This is just a speculative shot in the dark, but remember that the >>> domain of the From URI is important, and that the authentication "realm" >>> (domain) is part of the authentication credentials. So, what you have in >>> your 'fromdomain' and 'host' settings on the peer does matter. >>> >>> -- >>> This message was painstakingly thumbed out on my mobile, so apologies for >>> brevity, errors, and general sloppiness. >>> >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 260 Peachtree Street NW >>> Suite 2200 >>> Atlanta, GA 30303 >>> Tel: +1-678-954-0670 >>> Fax: +1-404-961-1892 >>> Web: <http://www.evaristesys.com/>http://www.evaristesys.com/ >>> >>> On Sep 30, 2011, at 3:16 AM, "cnasterisk" <[email protected]> wrote: >>> >>> hi, >>> Dear all. >>> I setted a sip account on a sip trunk. when a client call via this >>> sip trunk, asterisk call failed on this trunk. >>> I have captured the sip messages on the host where asterisk located, and >>> found that: >>> >>> 1. asterisk send a INVITE message to remote sip proxy without >>> "proxy-authorization" field. >>> 2. the remote sip proxy send back a >>> " SIP/2.0 407 Proxy Authentication Required" message. >>> 3. asterisk send a INVITE message with "proxy-authorization" field. >>> 4. remote proxy send back a "403(Forbidden)" message， that is mean "wrong >>> password" >>> >>> I also tested the sip account on a softphone, it works normal! >>> >>> why this happed？ and how can i solve it? >>> >>> >>> >>> 2011-09-30 >>> ------------------------------ >>> kevin >>> >>> -- >>> _____________________________________________________________________ >>> -- 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> >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> <http://lists.digium.com/mailman/listinfo/asterisk-users> >>> 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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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