> On March 16, 2015, 8:57 a.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? :-) > > rnewton wrote: > 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? >
We could always ask Allison :-) - Matt ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4488/#review14698 ----------------------------------------------------------- On March 13, 2015, 9:32 a.m., rnewton wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4488/ > ----------------------------------------------------------- > > (Updated March 13, 2015, 9:32 a.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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