Re: enabling dialing by sip uri

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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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