Test install tomorrow, need some advice

This last week I have been playing with Asterisk to eventually replace our current phone line. Today I brought two Cisco business IP phones today for a test (production) install of Asterisk tomorrow and to give a demonstration.

But i’m not sure on the best way of setting up my extentions.conf. Basicly when the Asterisk box gets a call from the SIP trunk it Dial()'s the main SIP phone (100) it gets a call for 15 seconds, if busy or no answer it Dail()'s SIP phone 101,102,103 at the same time. If no answer then go to a main voicemail. But each of my SIP phones in the config have their own voicemail, so when asterisk phone calls each of the other phones because the main phone is busy will it also run the voicemail() on each of the SIP clients? I’m not sure if that makes sense?

Secondly… My SIP provider, sipgate, allows 10 calls at once. How do I get it to send say a second call on the number to a different phone if the main phone is busy?

Thirdly… On the Cisco IP phones how do they interact with Asterisk to transfer a call to another extension. Does the phone just dial the extension on behalves of the client or what? How do I set that up?

I know that’s loads of questions, but I have no system to test these things tonight and I just want to be prepared for tomorrow.

Thanks, David Maitland.


You can define in your context something like:

... exten=>yourexten,1,Dial(SIP/100,15,Tt) exten=>yourexten,n,Dial(SIP/1001&SIP/102&SIP/103,15,Tt) exten=>yourexten,n,Voicemail(yourMainvoicemail@default)

With the above explample if the peer 100 is busy or dont answer after 15 secs the next step dial the other phones, you can use that or use the DIALSTATUS

The IP phones have their transfer capability you dont configure anything if you want to use the asterisk features check the features.conf.


Ok when you say [quote]test (production) install of Asterisk tomorrow[/quote] and follow it with [quote]to give a demonstration[/quote] but then ask basic questions, its realy too late to expect the demo to go OK isnt it.

You need to set the system up in a lab work out how it works and then demo it. Making sure you now the ins and outs.

I would look at one of the Gui systems be that elastix or PIAF as they do the donkey work for you.


I agree that it is too early for a demo.

I would disagree about starting with a GUI. I think you should gain enough understanding of what happens under the hood in the GUI, then use the GUI as a productivity aid. We get a lot of people here who use GUIs but are unable to debug the problems they get or follow debugging instructions.

I would not advise starting with Cisco phones. Reports suggest that their SIP implementations are quirky.

Just an update…

Got all of the equipment delivered this morning, Cisco phones & new server. The Cisco phones are really nice. Well made and have some really cool features…

It took a few hours to get the routing plans right, but by the end of the day I was able to show my boss, call phones, transferring, multi lines and voicemail etc

Next week, ordering more phone, new dedicated internet connection for the SIP trunk. All good :smiley:

Thanks for your advice.