Guys
I've attached a 7970 How-To I wrote a while back.
This works with both head and 1.2beta2.
Can I suggest that anyone wanting to play with any SCCP phone sign on to
http://lists.berlios.de/mailman/listinfo/chan-sccp-users
Where the CHAN_SCCP list is pretty active (Sergio and the guys have done a
fantastic job moving this project forward).
Please note I WILL NOT reply to 7970 questions posted here (It's of no
interest to many people on this list - and traffic is getting a bit heavy)
but will on the CHAN_SCCP list.
Regards
Paul
---------------------------------------------------------
To get the 7970 phone working with the current release of CHAN_SCCP you need
to follow these instructions:-
Please note this is not a comprehensive How-To but is a work in progress, as
such it is subject to revision and change. It did however work for my
particular setup.
For the purposes of testing it is advisable to follow these instructions to
a new clean build of Asterisk. It is not recommended to install this to a
production environment.
1. Required files for the Cisco 7970:-
Cnu70.2-7-4-134.sbn
CVM70.2-0-0-112.sbn
Jar70.2-9-0-117.sbn
TERM70.7-0-1-0s.LOADS
TERM70.DEFAULT.loads
TERM71.DEFAULT.loads
These are the files I've used, however your naming conventions regarding the
version numbers may be slightly different.
2. Install a new copy of Asterisk Stable onto a new box, follow the
instructions here:-
http://www.voip-info.org/wiki-Asterisk+installation+tips
3. Install the latest copy of CHAN_SCCP onto the server, please note there
are TWO branches of the CHAN_SCCP project and the one used for this
installation is here:-
http://chan-sccp.berlios.de/
4. Install the files from 1. above into your TFTPBOOT directory.
5. Although elsewhere it states you need two files to be configured in
TFTPBOOT (SEP<mac>.cnf.xml and XMLDefault.cnf.xml) I found that if the
SEP<mac>.cnf.xml file was present the phone wouldn't register properly
with
the Asterisk server. I have not been able to identify why this is the case
and therefore my workaround is to use just the XMLDefault.cnf.xml file.
My XMLdefault.cnf.xml file is listed below.
<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName>xxx.xxx.xxx.xxx</processNodeName>
;
replace XXX with asterisk IP address
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation30006 model="IP Phone
7970"></loadInformation30006>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>
6. Ensure you have modified the modules.conf from /etc/asterisk as follows
noload => chan_skinny.so
load => chan_sccp.so # this seems to have been deprecated in the current
release - please try with and without
7. Edit the sccp.conf file in /etc/asterisk as follows
[general]
keepalive = 20
debug = 1
context = XXXXX ; Please insert the appropriate context you use for
your extensions.conf here
dateFormat = D.M.Y
bindaddr = xxx.xxx.xxx.xxx ; Please insert your asterisk IP address
here
port = 2000
disallow = all
allow = ulaw ; The codec to be used by the phone, higher has higher
priority
allow = alaw ; The codec to be used by the phone, higher has higher
priority
firstdigittimeout = 16 ; dialing timeout for the 1st digit
digittimeout = 8 ; dialing timeout for subsequent digits
[devices]
type = 7970 ; device type (see below)
autologin = XYZ1,XYZ2,XYZ3 ; replace XYZ with the name you want to
call the lines (see below)
description = YYYYYYYY ; name displayed on the phone
tzoffset = +1 or -1 ; enter value to set correct timezone
transfer = 1
speeddial = XXXX,name
device => SEP<mac>
[lines]
id = 1000 ; future use
pin = 1234 ; future use
label = zzz ; label for line to appear on the phone
context = XXXXX ; insert the appropriate context you use for your
extensions.conf here
incominglimit = 2 ;more then 1 incoming line adds call hold
transfer = 1 ;per line transfer capability
mailbox = AAAA ; number of the mailbox you want calls to divert to on
no answer
vmnum = BBBB ; Number you want to dial when you hit the voicemail
button on phone (envelope)
cid_name = "DDDD" ; name you want assigned to called id variable
for this
line
cid_num = EEEE ; number you want assigned to caller id variable for
this line
line => XYZ1 ; The line you wish this variable set applied to (see
autologin in [devices])
---------------------------------------------------------------------------
If you wish to have more then one device specified then use the following
format:-
[devices]
Variables as per above example
Line space
Line space
Line space
Repeat variables for new device with a new SEP<mac> number for separate
device
If you wish to have more then one line per device (or multiple lines over
multiple devices):-
[lines]
Variables as per above example
Line space
Line space
Line space
Repeat variables as per above example with a line => XYZ1 for each autologin
specified in all the devices fields
--------------------------------------------------------------------------
I'm assuming that you have a working knowledge of configuring your
extensions.conf, dialplans, configuring DHCP, configuring TFTPBOOT etc...,
if you don't please read the asterisk wiki as it's outside the scope of
this
document.
Once you've done the above and started up asterisk plug your 7970 into your
network and add power, as the phone powers up BEFORE the speaker light goes
out press the # key (the phone lights should start to flash orange in
sequence), then on the handset press 123456789*0#. This causes the phone to
do a factory reset and should load the new files and settings from the
BOOTTFTP directory. If you've done it all correctly (and I've left
nothing
out) you should have a working handset, on the asterisk console you should
see the phone register with the asterisk server.