[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 55-58
> > <https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line55>
> >
> >     You have a Gosub here without a Return. That will unbalance the stack.

Where is the Return needed? On the h or o extensions? I'm still used to macros so I'm not skilled with the ol' Gosub yet.


> On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
> > /branches/13/configs/basic-pbx/extensions.conf, lines 64-67
> > <https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line64>
> >
> >     If this is a subroutine, it needs to Return().

It Returns on the other end of the Goto operations. Is that bad?


> On March 16, 2015, 1:57 p.m., Matt Jordan wrote:
> > /branches/13/configs/basic-pbx/extensions.conf, line 67
> > <https://reviewboard.asterisk.org/r/4488/diff/1/?file=72117#file72117line67>
> >
> >     Since your 'dialed-${DIALSTATUS}' extensions are subroutines, this needs to be invoked as a subroutine.

Ah, I thought Goto could be used within the operation of a Gosub and that any Return would Return out of the current Gosub. I probably misunderstood something fundamental about Gosubs. 


- rnewton


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4488/#review14698
-----------------------------------------------------------


On March 13, 2015, 2:32 p.m., rnewton wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4488/
> -----------------------------------------------------------
> 
> (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
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20150317/97ccfa12/attachment.html>