21 dec 2009 kl. 12.00 skrev jonas kellens: > My SIP-provider sends my a SIP-invite like this : > > INVITE sip:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c > Max-Forwards: 70 > From: <sip:[email protected]>;tag=f395877e02bf8eb2fd8f5a0e > To: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 1 INVITE > User-Agent: SysMaster VoIP Gateway v1.2.0 > Contact: <sip:[email protected]:5060> > Remote-Party-ID: > <sip:[email protected]>;party=calling;screen=yes;privacy=off > Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE > Content-Type: application/sdp > Content-Length: 261 > > This is my sip.conf : > > [outgoing] > type=peer > host=sip.XXX.tld > username=329298xxx6 > secret=my-secret > fromuser=329298xxx6 > disallow=all > allow=gsm > allow=alaw > > ; incoming > [329298yyy6] > type=user > host=sip.XXX.tld > context=mycontext > disallow=all > allow=gsm > allow=alaw > > The call does not come into the context "mycontext" but into the default > context... > > How can I authenticate this call so that it does not go into the default > context ???
Well, you have type=user on the incoming context, so that won't be matched for incoming calls. Change it to type=peer and you will be a much happier Asterisk user. /O _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
