Some time ago I posted a long guide for setting Cisco phones up. I wanted to post a quickstart guide for the 3PCC phones specifically, so here you go:
Note: 3PCC/Multiplatform firmware only applies to the Cisco 78xx and 88xx phones with the exception of the 8875 which runs PhoneOS. The 98xx series also runs PhoneOS. PhoneOS is a different setup procedure.
This is for version 12 of MPP firmware. Use chan_pjsip channel driver.
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Factory reset the phone. The general process is - hold # down, plug in phone, wait 30 seconds, then let go of # and press 123456789*0#
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Wait around 2-3 minutes with the phone plugged into a DHCP server. Eventually, the phone will show “Enter Activation Code” This is normal and will go away once the phone is provisioned.
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Press the Gear button on the phone and select Network Configuration, IPv4 Address settings, Connection Type and the phone’s IP will be displayed. Press Back to back out of this back to the activation screen
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From a web browser access the IPv4 address. Set the admin and user passwords if asked
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Access IPv4 address again and login. Click in the upper right until you are in Administrator and Advanced mode. (these are toggles)
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Check firmware version - Info, Status. As of 10/29 the most current firmware version is 12-0-8MPP
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Click the Voice, Ext 1 tab & fill it out from the extension page for the extension you want to use in FreePBX
a) Scroll down to Subscriber Information, fill out Display Name, userID, auth ID, password
b) Scroll up to Proxy and Registration, fil out Proxy: (ip address of your FreePBX server or hostname)
d) DialPlan change from (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) to
(*xx|[3469]11|0|00|[2-9]xxxxxx|*9xxxxxxxxxx|5xxx|3xxx|7xxx|8xxx|2xxx|4xxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) (this is for 4 digit internal extensions, modify to suit) -
Scroll down, Submit All Changes, After phone reboots make sure it registers
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Login to the phone again via web browser and make sure your admin, advanced
a) voice, SIP RTP Port Max change from 16482 to 32767 Submit All Changes (or leave this alone, it’s up to you - you want to make sure your translator and firewall agree here)
c) voice, Phone General station name extension, display name (use DID or extension #) VM number *98
d) Voice, phone, Multiple Paging Group, in the Group 1 Paging Script put
pggrp=224.168.168.168:34560;name=All;num=400;listen=yes; Submit All Changes -
Login to the phone again via web browser and make sure your admin, advanced
a) Voice, Regional, Time, Time Zone to GMT-8, change daylight savings to start=3/-1/7/2;end=11/-1/7/2;save=1
b) Voice, User, Screen Back Light Timer, Always ON, Screen Saver Enable no, screen saver clock UNLESS PHONE IS IN VERY BRIGHT AREA THEN TURN OFF BACKLIGHT
c) Voice, Att Console, Server Type Asterisk. Submit All Changes
Now the first extension is created. You might want to change SIP from UDP to TCP depending on your network or make other changes with names, text, etc. Unfortunately there’s a few bugs like Display Name isn’t the display name it’s the extension, etc.
For BLF keys:
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Login as Admin, Advanced
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Click Voice, ATT Console, set BXfer on Speed Dial Enable to Yes also make sure Asterisk is server type
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Click Voice, Phone and select an unused line key button (Line Key 2 for example)
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Change Extension to Disabled on that line key
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in Extended Function under this put in for example:
fnc=blf+sd;sub=821@$PROXY;ext=821;nme=Sales Desk x821
This will BLF monitor extension 821
Click Submit All Changes
BLF info:
Note that it is section 2 of that article.
Note that instead of BLF you can set Max Contacts to whatever number in pjsip for the extension and just register an additional line key on the phone into the extension you want to share.
Note that 3PCC phones support BLF URI which is another way of configuring BLF, information about that is here: