Sure thing,
First of all - I don't configure the phone manually. I use tftp config
files to accomplish this for the Cisco phones (I bring some home from
work... and don't want to load the firmware and configs manually
everytime!).
Here's a couple of links to get this done:
http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960
http://www.wheely-bin.co.uk/cisco/
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
-- TFTP server files (used to load & config phones) --
SIPDefault.cnf
# SIP Default Generic Configuration File
# Image Version
# image_version: P0S30202
image_version: "P0S3-06-3-00"
# Proxy Server
proxy1_address: "10.20.30.129" ; Can be dotted IP or FQDN
proxy2_address: "10.20.30.129" ; Can be dotted IP or FQDN
proxy3_address: "10.20.30.129" ; Can be dotted IP or FQDN
proxy4_address: "10.20.30.129" ; Can be dotted IP or FQDN
proxy5_address: "10.20.30.129" ; Can be dotted IP or FQDN
proxy6_address: "10.20.30.129" ; 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
# 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: "10.20.30.129" ; SNTP Server IP
Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or
directedbroadcast (default)
time_zone: PST ; 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: 0 ; 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: "" ; 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: 1 ; 0-Disabled (default), 1-Enabled
messages_uri: "8500"
logo_url: "http://10.20.30.129/asterisk/asterisk-tux.bmp"
directory_url: "http://10.20.30.129/asterisk/directory.html"
SIP<MAC ADDRESS).cnf (You need a config file for each telephone -- based
on MAC address)
# Line 1 appearance
line1_name: 3000
# Line 1 Registration Authentication
line1_authname: "3000"
# Line 1 Registration Password
line1_password: "cdplayer"
# Line 2 appearance
line2_name: 2003
# Line 2 Registration Authentication
line2_authname: "2003"
# Line 2 Registration Password
line2_password: "cdplayer"
####### New Parameters added in Release 2.0 #######
# All user_parameters have been removed
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Eldon Balzer" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Local 2002"
# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: "Local 2003"
####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP Phone" ; Limited to 15 characters (Default -
SIP
Phone)
# Phone Password (Password to be used for console or telnet login)
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
dialplan.xml (not needed but helps for speeding up the dialing)
<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 -->
<TEMPLATE MATCH="9,1.........." Timeout="0"
User="Phone"/> <!-- Long
Distance -->
<TEMPLATE MATCH="9,.........." Timeout="0"
User="Phone"/> <!--
Local numbers -->
<TEMPLATE MATCH="8,......." Timeout="0"
User="Phone"/> <!--
Corporate Dial plan-->
<TEMPLATE MATCH="2..." Timeout="0"
User="Phone"/> <!--
Corporate Dial plan-->
<TEMPLATE MATCH="3..." Timeout="0"
User="Phone"/> <!--
Corporate Dial plan-->
<TEMPLATE MATCH="8..." Timeout="0"
User="Phone"/> <!--
Corporate Dial plan-->
<TEMPLATE MATCH="1888......." Timeout="0"
User="Phone"/> <!-- IAX2
Dial plan to IAXTEL-->
<TEMPLATE MATCH="1800......." Timeout="0"
User="Phone"/> <!-- IAX2
Dial plan to IAXTEL-->
<TEMPLATE MATCH="1877......." Timeout="0"
User="Phone"/> <!-- IAX2
Dial plan to IAXTEL-->
<TEMPLATE MATCH="1866......." Timeout="0"
User="Phone"/> <!-- IAX2
Dial plan to IAXTEL-->
<TEMPLATE MATCH="*" Timeout="15"/>
<!--
Anything else -->
</DIALTEMPLATE>
RINGLIST.DAT (to have all those cool ring tones you've made!)(not necassary)
***each of the filenames have to be in the tftp servers root directory!
Ahh! ahh.pcm
Doh! doh.pcm
Are You There M AreYouThere.raw
ClockShop ClockShop.raw
Curley Curley.raw
Drums 1 Drums1.raw
Drums 2 Drums2.raw
FilmScore FilmScore.raw
FlintPhone FlintPhone.raw
HarpSynth HarpSynth.raw
Jamaica Jamaica.raw
Klaxons Klaxons.raw
KotoEffect KotoEffect.raw
MusicBox MusicBox.raw
Neuro Neuro.raw
Ohno Ohno.raw
Piano 1 Piano1.raw
Piano 2 Piano2.raw
Pop Pop.raw
Saxaphone 1 Sax1.raw
Saxaphone 2 Sax2.raw
Asleep asleep.raw
Caramba caramba.raw
MayIHelp mayihelp.raw
Dilbert Boss SICA-dilbert-BungeeBoss.raw
Dilbert Meeting SICA-dilbert-PHB.raw
Merlin2 merlin2.pcm
Merlin3 merlin3.pcm
Merlin4 merlin4.pcm
Merlin5 merlin5.pcm
Merlin6 merlin6.pcm
Merlin7 merlin7.pcm
And finally the asterisk side...
sip.conf
[2003]
type=friend
host=dynamic
username=2003
secret=cdplayer
context=default
nat=no
callgroup=2
pickupgroup=1
mailbox=2003
[3000]
type=friend
host=dynamic
username=3000
secret=cdplayer
context=default
nat=no
callgroup=2
pickupgroup=1
mailbox=3000
Let me know if you need anything else! For me this was relatively easy
as I've been working with Cisco's VoIP at the office... alot of the same
files are used with their SCCP firmware.
-= EB =-
> Hi,
>
> Could I ask how you have your sip.conf configured for the phones please.
>
> I am having probs getting mine to talk to asterisk........
>
> I am assuming its a manual config on the phones too, what do you put into
> the phones to get it to talk to Asterisk
>
> Many Thanks in advance
>
> Paul
> ----- Original Message -----
> From: <stuff@balzer.ca>
> To: "Asterisk Asterisk" <asterisk-users@lists.digium.com>
> Sent: Sunday, December 12, 2004 10:47 AM
> Subject: [Asterisk-Users] Caller ID info ZAP --> SIP??
>
>
>> Hi everyone,
>>
>> I've been toying with * for quite some time now. I've got two
Cisco
>> 7940's with the SIP firmware playing nice with *. I can also make
>> outbound calls via IAXTel (toll-free calls only) and all other calls I
>> have routed out my X100P-clone adapter.
>>
>> Here's my question... Is there a way to capture the inbound
callerid
>> from
>> my phone line (coming in on the X100P) and have it appear on my SIP
>> phones
>> properly?
>>
>> Or maybe I'm just doing this wrong. I want all my IP phones to
ring
>> when
>> someone from the outside calls my number. When this fails... send them
>> to
>> voicemail. Right now I just dial my IP phone when the outside line
>> rings.
>> If there is a different way about doing this -- please let me know!
>>
>> This is how my test config is right now:
>>
>> extensions.conf
>> [incoming]
>> exten => s,1,Dial(SIP/2002,20)
>>
>> The phone does ring... but for caller id info it just shows
"asterisk".
>>
>> Any help would be appreciated.
>>
>> -= EB =-
>> _______________________________________________
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>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>