Asus TS300-E4 PA4- User Manual

Asus TS300-E4 PA4

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Table of Contents:

  • Page 3 – Extensions using Dialplan pattern Matching
  • Page 4 – Tuning the system; Call waiting was turned off for each extension in sip.conf:
  • Page 5 – A context for blind transfers is provided and the
  • Page 6 – The immediate keyword is set in zapata.conf so that the; Managing Day, night, and Holiday Mode changes
  • Page 7 – open for business on Easter.; M O D E; Training users
  • Page 8 – is an Asterisk system right for your site?
  • Page 9 – conclusion; The Asterisk beginner would do well to start with the book; fone*CLI> core show application ApPlIcAtIoN not case sensitive; Administrator’s Guide for; Administrator’s Guide for the SoundPoint IP/SoundStation
  • Page 10 – Asterisk: The Future of
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A 3 6 - u s e r A s t e r I s k I N s tA L L At I O N

31

R o b e R T s o l o M o n

a 36-user Asterisk

installation

Phone System Administrator is one of several hats

Bob Solomon wears at a medium-sized non-profit in

New York City. He would like to participate in more

Asterisk projects.

[email protected]

m y n o n - p r o f i t e m p l o y e r n e e d e d

to replace an aging Toshiba KSU system

because the phones, some of which date

back to the ’80s, were falling apart and the

voicemail’s hard drive was grinding loudly.

I was able to convince management to

replace the system with an Asterisk-based

PBX by promising a lower end cost and the

elimination of vendor lock-in. In this article,

I walk you through the process I followed:

selecting phones, modifying the Asterisk

configuration, and training users.

Asterisk installations of the sort I had done previ-
ously, where Asterisk functioned as an answering
machine or a small office phone system, have been
documented in many places. This installation of-
fered an opportunity to take my Asterisk skills to
the next level. The system supports eight lines, 24
extensions, and over 36 voicemail users.

selecting phones

Originally, I proposed a system utilizing a chan-
nel bank and Aastra 9116 phones. The final design
used Polycom IP430 and IP601 SIP phones. The
keypad and voice quality of a sample 9116 was dis-
appointing in comparison to our existing Toshiba
EKT phones.

Business-quality analog phones are available at
about the same price point as the Polycom SIP
phones that we ultimately selected. To support the
SIP phone the only hardware required is an Ether-
net or, ideally, Power Over Ethernet port. The per
port cost of POE is much less than that of an FXS
(analog phone station) port.

The existing phone system provided voice-first in-
tercom; that is, intercom calls were answered by
the recipient’s phone on speaker phone, without
any action on the part of the recipient. A warning
tone preceded the call so that the recipient knew
when he or she no longer had privacy. An all-page
feature similar to voice-first intercom, but con-
necting to most phones on the system at once, one
way, was also provided. A conversation with man-
agement confirmed the importance of retaining
these features. I tested a sample Polycom IP430.
This phone could handle voice-first calls with bet-
ter sound quality than the existing system. The
keypad feel and sound quality compared favorably
with our existing phones.

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Summary

Page 3 - Extensions using Dialplan pattern Matching

; L O G I N : O c t O b e r 2 0 0 8 A 3 6 - u s e r A s t e r I s k I N s tA L L At I O N 33 since the memory was replaced. Please note that the symptoms of this mem-ory failure were freezes during disk I/O and md5sum sometimes reporting bad check sums on good files. The server was temporarily moved...

Page 4 - Tuning the system; Call waiting was turned off for each extension in sip.conf:

34 ; L O G I N : v O L . 3 3 , N O . 5 The following dialplan excerpt handles intercom calls placed from an inside context: ; line below is for voicemail for calling selfexten => _ [1-8]X,1,GotoIf($ [$ {CALLERID(num)} = $ {EXTEN}]?CheckVoiceMail); check if there is an extension if not go to voice...

Page 5 - A context for blind transfers is provided and the

; L O G I N : O c t O b e r 2 0 0 8 A 3 6 - u s e r A s t e r I s k I N s tA L L At I O N 35 <alertInfo voIpProt.SIP.alertInfo.1.value=”Ring Answer” voIpProt.SIP.alertInfo.1.class=”4”/><RING_ ANSWER se.rt.4.name=”Ring Answer” se.rt.4.type=”ring-answer” se.rt.4.timeout=”1000” se.rt.4.ringer=...

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