Difference between revisions of "Cisco 79xx"

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(update/clean on Cisco phone info)
(Merged Power over Ethernet (PoE) into Telephone system:Cisco PoE hack)
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* type {{k|3}}{{k|2}}{{k|2}}{{k|8}}{{k|7}}{{k|3}}{{k|7}}{{k|3}}{{k|8}}{{k|#}} (factreset#) and confirm with {{k|*}}
 
* type {{k|3}}{{k|2}}{{k|2}}{{k|8}}{{k|7}}{{k|3}}{{k|7}}{{k|3}}{{k|8}}{{k|#}} (factreset#) and confirm with {{k|*}}
 
* power cycle or press {{k|*}}, {{k|0}} and {{k|softkey4}} again
 
* power cycle or press {{k|*}}, {{k|0}} and {{k|softkey4}} again
 
=== Power over Ethernet (PoE) ===
 
 
This part gives some quick information on converting/applying 802.3af PoE for usage on the 79x0 series pre-standard PoE <abbr title="Powered Device">PD</abbr>s.
 
 
(needs verification)
 
swap blue and brown, and for those pairs: swap color and white
 
 
For a type B <abbr title="Power Sources Equipment">PSE</abbr> uses the spare pairs (4,5 and 7,8) with being 802.3af: undetermined polarity.
 
Test power by placing a 25kohm resistor between pins 4 and 7.
 
Now, make sure the pins 4 and 5 contain the positive (+48) voltage, while pins 7 and 8 have the negative (GND) voltage.
 
You might need to swap pairs 4+5 with 7+8 (blue with brown)
 
TODO: not sure if the resistor is needed after testing; maybe it's needed to swap the colors per pair
 
 
For a type A PSE (like NetGear FS108P), make sure the switch supports <abbr title="also known as 'auto-MDIX'">auto-sensing MDI/MDIX</abbr>
 
Test power by placing a 25kohm resistor between pins 1 and 3.
 
Grab a crossover cable and make sure the voltage polarity is correct.
 
 
source: http://www.voip-info.org/wiki/view/Cisco+POE
 
  
 
== phones==
 
== phones==

Revision as of 17:32, 26 October 2015


Marked as outdated
needs work/cleaning
Cisco IP Phone 7940 series

synopsis

Here is some various information of the Cisco IP 7900 series phones and extension modules as used in our Cisco phone deployment. Currently, there are only two 7940 in the space, both with line 2 configured as an auto-answer intercom.

current dialplan

The current dialplan is set up so that you can directly dial external numbers (if we have pre-paid credit), and dial internal service numbers and extensions. Additional entries are needed for star(*) services which requires the user to postfix with a pound(#) sign.

<DIALTEMPLATE>
    <TEMPLATE MATCH="\*..#"            Timeout="0" User="Phone"/> <!-- 2 digit activate -->
    <TEMPLATE MATCH="\**#"             Timeout="3" User="Phone"/> <!-- 2 digit activate param -->
    <TEMPLATE MATCH="\*...#"           Timeout="0" User="Phone"/> <!-- 3 digit activate -->
    <TEMPLATE MATCH="\**#"             Timeout="3" User="Phone"/> <!-- 3 digit activate param -->
    <TEMPLATE MATCH="\*..#"            Timeout="0" User="Phone"/> <!-- 2 digit activate -->
    <TEMPLATE MATCH="\*#..#"           Timeout="0" User="Phone"/> <!-- 2 digit status -->
    <TEMPLATE MATCH="\*#...#"          Timeout="0" User="Phone"/> <!-- 3 digit status -->
    <TEMPLATE MATCH="#..#"             Timeout="0" User="Phone"/> <!-- 2 digit cancel -->
    <TEMPLATE MATCH="#...#"            Timeout="0" User="Phone"/> <!-- 3 digit cancel -->
    <TEMPLATE MATCH="00*"              Timeout="6" User="Phone"/> <!-- International calls-->
    <TEMPLATE MATCH="0........."       Timeout="1" User="Phone"/> <!-- National calls -->
    <TEMPLATE MATCH="10.."             Timeout="0" User="Phone"/> <!-- Local extensions -->
    <TEMPLATE MATCH="12.."             Timeout="0" User="Phone"/> <!-- Customer service -->
    <TEMPLATE MATCH="13.."             Timeout="0" User="Phone"/> <!-- Services -->
    <TEMPLATE MATCH="131.........."    Timeout="1" User="Phone"/> <!-- Services -->
    <TEMPLATE MATCH="#31#.........."   Timeout="1" User="Phone"/> <!-- Services -->
    <TEMPLATE MATCH="\*31\*.........." Timeout="1" User="Phone"/> <!-- Services -->
    <TEMPLATE MATCH="132.........."    Timeout="1" User="Phone"/> <!-- Services -->
    <TEMPLATE MATCH="\*10.."           Timeout="0" User="Phone"/> <!-- Intercom -->
    <TEMPLATE MATCH="\*19.."           Timeout="0" User="Phone"/> <!-- Intercom -->
    <TEMPLATE MATCH="19.."             Timeout="0" User="Phone"/> <!-- Remote extensions -->
    <TEMPLATE MATCH="2222"             Timeout="0" User="Phone"/> <!-- Emergency -->
    <TEMPLATE MATCH="4444"             Timeout="0" User="Phone"/> <!-- Emergency -->
    <TEMPLATE MATCH="112"              Timeout="0" User="Phone"/> <!-- Emergency -->
    <TEMPLATE MATCH="911"              Timeout="0" User="Phone"/> <!-- Emergency -->
    <TEMPLATE MATCH="999"              Timeout="0" User="Phone"/> <!-- Emergency -->
    <TEMPLATE MATCH="*"                Timeout="4" User="Phone"/> <!-- All Other -->
</DIALTEMPLATE>

various

Unlock settings menu =

SCCP: **#
SIP: settings9 password

Reboot/reload phone

SCCP (reload/hot boot): **#**
SIP (lukewarm boot):*+6+settings

Or just pull the power or 'power over ethernet' plug and wait a second

Do a factory default

Note: couldn't get it to work on 7910

794x(G)/796xG:

  • press and hold # while cold booting (before or when the green headset lights)
  • release it after the red mute lights and dims
  • phone says 'Reset key sequence detected' or the like
  • type 123456789*0#
  • save config: 1=yes, 2=no (choose 2 to clear everything)

7905:

  • press and hold *, 0 and softkey4
  • type 322873738# (factreset#) and confirm with *
  • power cycle or press *, 0 and softkey4 again

phones

Tested all Cisco equipment with an old unregulated printer power supply of 30V= (measured 39.6V). 18V unregulated (24.6V measured) does not power them at all.

7905

SIP phone, fast boot time. Currently, it is unknown where it gets its current dialplan from, but you can force a call by ending with #.

7910

  • 7910 phones only work with the sccp firmware
  • the 6 softkeys start at button position 5 in the FS config

all phones

  • To make sure sip firmware loaded phones work across the double-NAT openVPN tunnel setup, apply the following settings (either via the config file or the unlocked menu):
    • Register Expires = 100 (shorter than Linux' UDP 'establised' state time, which is 120 seconds)
    • Outbound proxy = same as Proxy (not sure)
    • NAT enabled = yes
    • NAT address = 10.0.50.13
      all this causes the extension to register with contact: "user" <sip:10xx@10.0.50.13:xxxx> and enables receiving calls
  • sccp firmware loaded phones have issues with RTP audio over the double-NAT openVPN tunnel setup (signaling works).
    The only way tested to work is to have a local FreeSWITCH box handling the phone traffic.

7914

  • 7914 expansion panels only work with sccp firmware loaded phones
  • the data cable is a 1:1 RJ12 connection with the first unit being powered by the phone (second is yet unknown)
  • the RS232/AUX RJ12 pinout (pin 1 is leftmost) (pinout needs verification):
pin function
1 +48V in (phone: +48v out)
2 RxD (phone: TxD)
3 signal ground
4 TxD (phone: RxD)
5 unknown/NC
6 power ground

The serial settings should be 9600/8-N-1, signal levels are unknown (presumably ±12v). Also, the protocol is unknown.

Tested the 7914 with with 7940 and 7960: the buttons on the extension board start with button position 3 and 7 respectively in the FS config.