thats what i am passing exten=> 456,1,Dial(SIP/adf:[EMAIL PROTECTED]:9060
adf is username and 123 is the password On Wed, Sep 17, 2008 at 6:01 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: > > 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 > -- 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
