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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4488/">https://reviewboard.asterisk.org/r/4488/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By rnewton.</div>
<p style="color: grey;"><i>Updated March 24, 2015, 9:53 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Addressed all findings, modified a few comments here or there.
Tested with 4503 and 4504. A quick summary of the testing I ran through. I tested all the requirements for all four patches combined.
*patch1 - internal stuff*
internal user to internal user (audio): PASS
internal user to internal user (voicemail-unavail): PASS
internal user to internal user (voicemail-busy): PASS
internal user can check voicemail: PASS
deskphone displays MWI indication: PASS
*patch2 - outside connectivity*
registration to ITSP(DCS) comes up: PASS
internal user dials out ITSP with 7 digit number: PASS
internal user dials out ITSP with 10 digit number: PASS
internal user dials out ITSP with 10+1 digit number: PASS
internal user dials main IVR internally: PASS
restricted number patterns work successfully: PASS
inbound calls reach the main IVR: PASS
external user can reach external voicemail feature via DID: PASS
external users can dial internal users directly via DID match: PASS
*patch3 - queues with external and internal access*
sales queue reached internally: PASS
externally: PASS
sales queue rings Terry, Garnet and Franny in ring-all: PASS
customer advocate queue reached internally: PASS
externally: PASS
customer advocate Queue rings Maria, Dusty and Tommie in ring-all: PASS
*patch4 - conferences*
employee conference can be dialed by internal users: PASS
at least two parties in employee conference with audio: PASS
customer conference can be dialed into by internal user and transfer in external users: PASS
at least two parties, including an external party in customer conference with audio: PASS
*ALL PATCHES COMBINED*
All IVR options go to the correct feature/extension: PASS
CDR Master.csv does not record any intra-office calls: PASS
CDR Master.csv records calls to/from the ITSP account: PASS</pre>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>/branches/13/configs/basic-pbx/pjsip.conf <span style="color: grey">(433333)</span></li>
<li>/branches/13/configs/basic-pbx/modules.conf <span style="color: grey">(433333)</span></li>
<li>/branches/13/configs/basic-pbx/logger.conf <span style="color: grey">(433333)</span></li>
<li>/branches/13/configs/basic-pbx/extensions.conf <span style="color: grey">(433333)</span></li>
<li>/branches/13/configs/basic-pbx/cdr_custom.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/branches/13/configs/basic-pbx/cdr.conf <span style="color: grey">(PRE-CREATION)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4488/diff/" style="margin-left: 3em;">View Diff</a></p>
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