[asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!

[rnewton]

> On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
> > /branches/13/configs/basic-pbx/extensions.conf, lines 135-136
> > <https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line135>
> >
> >     I'm assuming we're going to replace the prompts eventually? :-)

Yeah I wasn't sure if we wanted to deliver some custom sounds with this example or just placeholders. It would be nice if we had some custom sounds to go with it.

If we did, where would be the best place for the custom, example-specific sounds to live in the source?

Who do we have record the sounds? A professional? Or just me?


> On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
> > /branches/13/configs/basic-pbx/modules.conf, line 25
> > <https://reviewboard.asterisk.org/r/4488/diff/1/?file=72119#file72119line25>
> >
> >     Don't you need a config file for this?

Forgot to 'svn add' cdr.conf and cdr_custom.conf. Whoops. 


- rnewton


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On March 13, 2015, 2:32 p.m., rnewton wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4488/
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> 
> (Updated March 13, 2015, 2:32 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Howdy, here is another patch for the Super Awesome Company configuration. We are still in phase 1. The general requirements are posted on the wiki: https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company
> 
> The specific requirements this patch meets are below:
> 
> pjsip.conf
> 
>  * SIP ITSP configuration example and have place holders for the required authentication bits.
>  ** Assume that Asterisk does not have a public IP address, and sits behind a NAT with its desk phones.
>  * Have outbound registration to the SIP trunk, and an endpoint that represents the SIP trunk.
>  * Inbound calls received from the SIP trunk should go into their own context.
> 
> extensions.conf
> 
>  * Match the outbound dial request so that it can only dial US area codes.
>  ** Don't let people dial 900 numbers, international numbers, or any other numbers that could result in a charge
>  * Inbound calls from the SIP trunk should hit a basic Auto Attendant that prompts them for the extension to dial, after greeting them to SAC.
>  * If an inbound call matches a DID that maps to a specific extension/device, dial that extension/device directly.
> 
> Billing
> 
>  * Make sure CDRs output all calls that are from/to the SIP trunk. These should be logged to a CSV.
>  * For intra-office calls, kill the CDRs.
> 
> Additional Requirements Noted:
> 
>  * For outbound calls, each SAC employee’s 10-digit DID number is provided as their Caller ID.
>  * Voicemail may be accessed remotely by employees who dial 256-555-1234. When employees dial voicemail remotely, they must input both their mailbox number and their pin code.
>  * 7, 10 and 10+1 digit dialing for local and long distance calls.
>  * Internal dialing of otherwise inbound features, 
>  ** 1100 to reach the main IVR.
>  * The IVR options possible without getting into Phase 2.
> 
> 
> Diffs
> -----
> 
>   /branches/13/configs/basic-pbx/pjsip.conf 432866 
>   /branches/13/configs/basic-pbx/modules.conf 432866 
>   /branches/13/configs/basic-pbx/logger.conf 432866 
>   /branches/13/configs/basic-pbx/extensions.conf 432866 
> 
> Diff: https://reviewboard.asterisk.org/r/4488/diff/
> 
> 
> Testing
> -------
> 
> Setup with a Digium Cloud Services trunk and a few internal phones.
> Internal to Internal calls.
> Calls Internal to voicemail and other features.
> External to internal DID calls.
> External to internal feature calls.
> 
> Basically tried to call as many ways as I could through all the various features. Everything seemed to work.
> 
> 
> Thanks,
> 
> rnewton
> 
>

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