If there is a secret= on the receiving peer, the sending peer needs to provide that secret. Along with a username.
Rizwan Hisham wrote: > Hi all, > I am having a problem with sip uri incoming calls. I have 2 asterisk > servers both are 1.4.2. <http://1.4.2.> i dial sip uri from one asterisk > server which sends the call to the other asterisk server by seeing its > domain name in the uri. Invite reaches the recieving asterist server but > the call is not autenticated. Everytime i see the following NOTICE on > the asterisk server (caller end) > > [Sep 17 15:38:24] NOTICE[4594]: chan_sip.c:11968 handle_response_invite: > Failed to authenticate on INVITE to '"rizwan" <sip:[EMAIL PROTECTED]:9860 > <http://sip:[EMAIL PROTECTED]:9860>>;tag=as089d4adb' > > My dialplan on caller end is: > > [directcall] > exten=> 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060 > <http://abc:[EMAIL PROTECTED]:9060>) > exten=> 123,2,Hangup() > > exten=> 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060 > <http://adf:[EMAIL PROTECTED]:9060>) > exten=> 456,2,Hangup() > > SIP general settings on receiving end are: > > [general] > context=uricall-incoming > allowoverlap=no > bindport=9060 > bindaddr=0.0.0.0 <http://0.0.0.0> > srvlookup=yes > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > relaxdtmf=yes > useragent=Asterisk PBX > dtmfmode = rfc2833 > nat=no > canreinvite=yes > > peer settings on receiving end: > > [adf] > username=adf > type=friend > secret=XXX > qualify=25000 > nat=yes > insecure=port,invite > host=dynamic > dtmfmode=rfc2833 > context=sipuri-incoming > canreinvite=yes > callerid="adf xyz" <123> > accountcode=6:0:adf > amaflags=default > disallow=all > allow=g729 > allow=ulaw > allow=alaw > allow=gsm > > am i doing something wrong here? > > > -- > Best Regards > Rizwan Hisham > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
