[asterisk-users] enabling dialing by sip uri

[Kevin P. Fleming]

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
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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