[asterisk-users] enabling dialing by sip uri
Arif Hossain
aftnix at gmail.com
Thu May 10 15:36:56 CDT 2012
On Thu, May 10, 2012 at 11:50 AM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> 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).
>
I'm using the asterisk's ip to form sip uri at the sip client. So it
resolves to asterisk no doubt.
> --
> 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
> Check us out at www.digium.com & www.asterisk.org
>
> --
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-aft
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