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).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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