[asterisk-users] enabling dialing by sip uri

[Kevin P. Fleming]

On 05/10/2012 03:36 PM, Arif Hossain wrote:
>> 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.
>

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.

-- 
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