running 2.6.19-gentoo-r5

asterisk-1.2.16.tar.gz
chan_sccp-20060408.tar

tar -xvzf asterisk-1.2.16.tar.gz
tar -xvzf chan_sccp-20060408.tar

cd asterisk-1.2.16
./configure
make 
make install
make samples

cd ../chan_sccp-20060408
make
make install

as basic as i could sip.conf / sccp.conf / extensions.conf

Asterisk server running on 192.168.1.245
SIP Phones MAC: 000000002000
SIP Phones Number: 2000
SCCP Phones MAC: 000000002001
SCCP Phones Number: 2001

##########
sip.conf
##########

[general]
port=5060
bindaddr=0.0.0.0
allow=all
;context=bogon-calls
context=from-sip

[2000]
type=friend
username=2000
secret=blah
context=from-sip
host=dynamic
disallow=all
allow=ulaw
mailbox=2000

###########
sccp.conf
###########

[general]
keepalive = 30

context = internal
dateFormat = D.M.YA
bindaddr = 192.168.1.245
port = 2000
debug = 0

[devices]
type        = 7914
description = Reception
tzoffset    = 0
autologin   = 2001
device => SEP000000002001

[lines]
id            = 2001
pin           = 1234
label         = Reception
description   = Reception
context       = from-sccp
incominglimit = 2
mailbox       = 2001
vmnum         = 8500
cid_name      = Reception
cid_num       = 2001
line => 2001

################
extensions.conf
################

[general]
static=yes 
writeprotect=yes 

[from-sip]
exten => 2001,1,Dial(SCCP/2001,30)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup

[from-sccp]
exten => 2000,1,Dial(SIP/2000,30)
exten => 2000,2,Voicemail(u2000)
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup

##############
modules.conf
##############

[modules]
autoload=yes
noload => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
noload => app_intercom.so
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
load => res_musiconhold.so
noload => chan_alsa.so
load => chan_sccp.so
noload => chan_skinny.so

[global]

; end files

For the sccp 7960 phone on the tftp server I had:

SEP000000002001.cnf.xml
XMLDefault.cnf.xml
P00305000301.bin
P00305000301.sbn
English_United_States/7960-font.xml
United_States/7960-tones.xml

For the 7914 phone on the tftp server I had:

S00104000000.bin
S00104000000.sbn
S00104000100.sbn

For the SIP 7960 phone on the tftp server I had:

P0S3-07-3-00.bin
P0S3-07-3-00.loads
P0S3-07-3-00.sb2
P0S3-07-3-00.sbn
RINGLIST.DAT
dialplam.xml
OS79XX>txt
SIP000000002000.cnf
SIPDefault.cnf

For the SEP files for the 7960 I had:

####################
XMLDefault.cnf.xml
####################

<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>192.168.1.245</processNodeName>
</callManager>
</member>
</members>
<loadInformation124 model="Addon 7914">S00104000000</loadInformation124> 
<loadInformation7  model="IP Phone 7960">P00305000301</loadInformation7>
</callManagerGroup>
</Default>

########################
SEP000000002001.cnf.xml
########################

<device>
<devicePool>
<callManagerGroup>
<members>
<member  priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>192.168.1.245</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<versionStamp>{Jan 01 2002 00:00:00}</versionStamp>
<addOnModules>
<addOnModule idx="1">
<loadInformation>S00104000000</loadInformation>
</addOnModule>
<addOnModule idx="2">
<loadInformation>S00104000000</loadInformation>
</addOnModule>
</addOnModules>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<idleTimeout>0</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL></servicesURL>
</device> 


Made directory English_United_States under /tftpboot and put empty file
7960-font.xml in it.

Made directory United_States under /tftpboot and put empty file
7960-tones.xml in it.

For the SIP files for the 7960 I had:

###############
SIPDefault.cnf
###############

# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-07-3-00
# Proxy Server
proxy1_address: "192.168.1.245"		; Can be dotted IP or FQDN
proxy2_address: ""		; Can be dotted IP or FQDN
proxy3_address: ""		; Can be dotted IP or FQDN
proxy4_address: ""		; Can be dotted IP or FQDN
proxy5_address: ""		; Can be dotted IP or FQDN
proxy6_address: ""		; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060 
proxy2_port: 5060 
proxy3_port: 5060 
proxy4_port: 5060 
proxy5_port: 5060 
proxy6_port: 5060 

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600 

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw
#preferred_codec: 729a, 711alaw, g711ulaw

# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: avt

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3
# SIP Timers
timer_t1: 500 			; Default 500 msec
timer_t2: 4000 			; Default 4 sec
sip_retx: 10			; Default 10
sip_invite_retx: 6 		; Default 6
timer_invite_expires: 180 	; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"		; Example:  ./sip_phone/

# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: ""			; SNTP Server IP Address
sntp_mode: directedbroadcast	; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EST			; Time Zone Phone is in
dst_offset: 1			; Offset from Phone's time when DST is in effect 
dst_start_month: April		; Month in which DST starts
dst_start_day: ""		; Day of month in which DST starts
dst_start_day_of_week: Sun	; Day of week in which DST starts
dst_start_week_of_month: 1	; Week of month in which DST starts
dst_start_time: 02		; Time of day in which DST starts
dst_stop_month: Oct		; Month in which DST stops
dst_stop_day: ""		; Day of month in which DST stops
dst_stop_day_of_week: Sunday	; Day of week in which DST stops
dst_stop_week_of_month: 8	; Week of month in which DST stops 8=last week of month
dst_stop_time: 2		; Time of day in which DST stops
dst_auto_adjust: 1		; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1		; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 0			; Default 0 (Do Not Disturb feature is off)

# Caller ID Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0		; Default 0 (Disable sending all calls as anonymous) 

# Anonymous Call Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0		; Default 0 (Disable blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101		; Default 101

# Sync value of the phone used for remote reset 
sync: 1				; Default 1

####### New Parameters added in Release 2.1 #######

# Backup Proxy Support
proxy_backup: ""		; Dotted IP of Backup Proxy
proxy_backup_port: 5060		; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "" 		; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060	; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0			; VAD setting 0-disable (Default), 1-enable

####### New Parameters added in Release 2.2 ######

# NAT/Firewall Traversal
nat_enable: 0                   ; 0-Disabled (default), 1-Enabled
nat_address: ""		        ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060      	; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 	; Start RTP range for media (default - 16384)
end_media_port: 32766   	; End RTP range for media (default - 32766)
nat_received_processing: 0	; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""	 	; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060       ; default is 5060

####### New Parameter added in Release 3.0 #######

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1		; 0-Disabled, 1-Enabled (default)

####### New Parameters added in Release 3.1 #######

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1	; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into the phone) 
telnet_level: 1			; 0-Disabled (default), 1-Enabled, 2-Privileged

####### New Parameters added in Release 4.0 #######

# XML URLs
services_url: ""		; URL for external Phone Services
directory_url: "http://www.telegenetic.net/~entropy/directory.xml" ; URL for external Directory location
logo_url: ""			; URL for branding logo to be used on phone display

# HTTP Proxy Support
http_proxy_addr: ""		; Address of HTTP Proxy server
http_proxy_port: 80		; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""              ; restricted to dotted IP
dyn_dns_addr_2: ""              ; restricted to dotted IP
dyn_tftp_addr: ""               ; restricted to dotted IP

# Remote Party ID
remote_party_id: 0		; 0-Disabled (default), 1-Enabled

####### New Parameters added in Release 4.4 #######

# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0		; Default 0 (Call Hold Ringback feature is off)

####### New Parameters added in Release 6.0 #######

# Dialtone Stutter for MWI 
stutter_msg_waiting: 0		; 0-Disabled (default), 1-Enabled

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0			; 0-Disabled (default), 1-Enabled

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: ""   ; Has no effect on SIP messaging


####################
SIP000000002000.cnf
####################

line1_name: 2000
line1_authname: "2000"
line1_password: "blah"
line1_displayname: "Test Phone"
phone_label: "Test Phone"
phone_prompt:   "SIP Phone"
phone_password: "cisco" 
user_info: none 

#############
OS79XX.txt
#############

P003-07-3-00

###############
dialplan.xml
###############


<DIALTEMPLATE>
    <TEMPLATE MATCH="0"              Timeout="1" User="Phone"/> <!-- Local operator-->
    <TEMPLATE MATCH="9,011*"         Timeout="6" User="Phone"/> <!-- International calls-->
    <TEMPLATE MATCH="9,0"            Timeout="8" User="Phone"/> <!-- PSTN Operator-->
    <TEMPLATE MATCH="9,11"           Timeout="0" User="Phone" Route="Emergency" Rewrite="9911"/>
    <TEMPLATE MATCH="w!"             Timeout="1" User="PHONE" Route="Emergency" Rewrite="9911"/>
    <TEMPLATE MATCH="9,.11"          Timeout="0" User="Phone"/> <!-- Service numbers -->
    T<TEMPLATE MATCH="9,1.........."  Timeout="0" User="Phone"/> <!-- Long Distance -->
    <TEMPLATE MATCH="9,......."      Timeout="0" User="Phone"/> <!-- Local numbers -->
    <TEMPLATE MATCH="2..."           Timeout="0" User="Phone"/> <!-- Corporate Dial plan-->
    <TEMPLATE MATCH="3..."           Timeout="0" User="Phone"/> <!-- Corporate Dial plan-->
    <TEMPLATE MATCH="*"              Timeout="15"/>             <!-- Anything else -->
    <TEMPLATE MATCH="123#45#6"       Timeout="0" User="Phone"/> <!-- Match `#' --> 
    <TEMPLATE MATCH="12\*345"        Timeout="0" User="Phone"/> <!-- Match * Char -->
    <TEMPLATE MATCH="8,...."         TIMEOUT="0" Tone="Bellcore-Hold" /> <!-- Play Hold -->
    <TEMPLATE MATCH="8,123,...."     TIMEOUT="0" Tone="Bellcore-Hold" Tone="Cisco-Zip" /> <!--Play Hold after 8, Play Zip Tone after 123-->
</DIALTEMPLATE>