[asterisk-users] how asterisk work with VoIP trunk?

[Steve Edwards]

On Thu, 9 Jun 2011, virendra bhati wrote:

> I mean when we call on DID then call will come to my server and then I 
> want to move this call to any SIP extension. But call will not come to 
> extension just got message "device not in use". But device already 
> registered into asterisk server.

I'm not an expert in SIP messaging. Hopefully, my description will not 
materially mislead you.

A SIP call is initiated by sending an INVITE. This starts a dialog where 
the 2 endpoints exchange messages to determine authentication, which 
codecs are available and which will be used, the ports to be used for RTP 
(audio), messages signaling 'in-call' DTMF, and messages requesting the 
termination of the call.

When a caller calls the DID you are renting, the termination provider will 
send your Asterisk server an INVITE. The provider knows where to send the 
INVITE either because your Asterisk server REGISTERed with the provider or 
because you told the provider the host name or IP address of your Asterisk 
server.

When your provider and your Asterisk server have agreed upon all the 
details, Asterisk will start executing your dialplan at the context 
matching the context specified in sip.conf. The extension within the 
context is taken from one of the SIP headers in the INVITE. Probably the 
INVITE header or the TO: header. The priority within the exten will be 1. 
There are extension pattern matching rules so you don't have to specify 
every possible extension.

When you want to forward the call to an extension, you execute the 'dial' 
application, specifying the destination endpoint which starts another SIP 
dialog similar to above and then Asterisk bridges the 2 channels.

HTH.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000