enabling dialing by sip uri

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Asterisk Users 3 Comments

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).

3 thoughts on - enabling dialing by sip uri

  • I’m using the asterisk’s ip to form sip uri at the sip client. So it
    resolves to asterisk no doubt.

  • You’ll have to provide more details (primarily a CLI log) then in order
    for anyone to be able to help you. You said that Asterisk “shows
    extension is rejected”, but extensions don’t get rejected. Extensions
    can be ‘not found’, but that’s very different from rejected.