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.

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Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       [email protected]      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

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