Hi all, I am having a problem with sip uri incoming calls. I have 2 asterisk servers both are 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 >;tag=as089d4adb' My dialplan on caller end is: [directcall] exten=> 123,1,Dial(SIP/abc:[EMAIL PROTECTED]:9060) exten=> 123,2,Hangup() exten=> 456,1,Dial(SIP/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 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
