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).
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