[asterisk-dev] [Code Review] 4488: Super Awesome Company: Phase 1 - Patch 2 - Outside Connectivity!
rnewton
reviewboard at asterisk.org
Thu Mar 26 17:27:23 CDT 2015
> On March 26, 2015, 3:37 p.m., Jonathan Rose wrote:
> > /branches/13/configs/basic-pbx/pjsip.conf, line 64
> > <https://reviewboard.asterisk.org/r/4488/diff/2/?file=72803#file72803line64>
> >
> > Was this intended to be commented out like this?
Nope, great catch. That is an artifact from testing the config. I forgot to uncomment it back. Thanks!
- rnewton
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On March 24, 2015, 9:53 p.m., rnewton wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4488/
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>
> (Updated March 24, 2015, 9:53 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 433333
> /branches/13/configs/basic-pbx/modules.conf 433333
> /branches/13/configs/basic-pbx/logger.conf 433333
> /branches/13/configs/basic-pbx/extensions.conf 433333
> /branches/13/configs/basic-pbx/cdr_custom.conf PRE-CREATION
> /branches/13/configs/basic-pbx/cdr.conf PRE-CREATION
>
> 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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