On Thu, May 10, 2012 at 11:50 AM, Kevin P. Fleming <[email protected]> wrote: > On 05/10/2012 09:39 AM, Arif Hossain wrote: >> >> I have following sip account : >> >> Name/username Host Dyn >> Forcerport ACL Port Status Description >> demo-alice/demo-alice 192.168.7.47 D >> N 1080 Unmonitored >> demo-bob/demo-bob 192.168.7.47 D >> N 5060 Unmonitored >> >> and i have set up the following extensions for them: >> >> ASTERISK_IP=192.168.7.39 >> >> [users] >> exten=>6001,1,Dial(SIP/demo-alice,20) >> exten=>6002,1,Dial(SIP/demo-bob,20) >> >> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled) >> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled) >> exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN}) >> exten => _.,n,HangUp()u >> >> [macro-uri-dial] >> exten=>s,n,NoOp(Calling as SIP address: ${ARG1}) >> exten=>s,n,Dial(SIP/${ARG1},60) >> >> >> But if i dial sip uri the call does not happen. asterisk cli shows >> extension is rejected. > > > Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone, > and that URI does not resolve to the Asterisk server as its target, then the > INVITE request sent by the phone should not even be sent to Asterisk at all > (it should go to wherever the URI resolves to). >
I'm using the asterisk's ip to form sip uri at the sip client. So it resolves to asterisk no doubt. > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- 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 -- -aft -- _____________________________________________________________________ -- 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
