Has anyone done this yet?
I have Cisco SIP 8.x loaded
How do I set this up?
I assumed I would place the IP address to my asterisk server in the tftp setting.
But it still does not work.
What am I doing wrong?
Hi I have about 50 7940s working with Asterisk SIP 8.6.
Here are the configs nothing special.
sip.conf (extention 316)
[316]
tos=reliability
disallow=all
allow=ulaw
allow=g729
mailbox=316
type=friend
username=316
callerid=316
host=dynamic
context=internal
canreinvite=no
secret=1234
nat=no
qualify=yes
callgroup=1
pickupgroup=1
########################
SipDefault.cnf
########################
SIP Default Generic Configuration File
Image Version
image_version: P0S3-08-6-00
Proxy Server
proxy1_address: “192.168.5.10” ; 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
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: “192.168.5.10” ; SNTP Server IP Address
sntp_mode: anycast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EAST ; 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: 0 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format : D/M/Y
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: “192.168.5.10” ; 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: 2 ; 0-Disabled (default), 1-Enabled, 2-Privileged
####### New Parameters added in Release 4.0 #######
; 0-Disabled (default), 1-Enabled
XML URLs
services_url: “http://192.168.5.10/services.xml” ; URL for external Phone Services
directory_url: “http://192.168.5.10/pcg_dir.xml” ; URL for external Directory location
logo_url: “http://192.168.5.10/pcg.bmp” ; 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
#Voice Mail extention
messages_uri: 8500
RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0
#Relase DHCP
dhcp_address_released: yes
#Transfer by hanging up the phone
transfer_onhook_enabled:1
#####################
SIPMACADDRESS.cnf
#####################
SIP Configuration Generic File
#user 316
Line 1 appearance
line1_name: 316
Line 1 Registration Authentication
line1_authname: “316”
Line 1 Registration Password
line1_password: “1234”
Line 2 appearance
line2_name:
Line 2 Registration Authentication
line2_authname: “”
Line 2 Registration Password
line2_password: “”
####### 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: “” ; Has no effect on SIP messaging
Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: “316”
Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: “”
####### 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: “123” ; Limited to 31 characters (Default - cisco)
User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
Does something need to be activated on the asterisk server to enable this?
I have a tftp server running on the Asterisk boxes with these files in it and my DHCP server tells/gives the phone the IP of the tftp server (option 66 ip 192.168.5.10). So the phone knows to contact the TFTP server 192.168.5.10 to get its config.
dialplan.xml
OS79XX.TXT
SIPDefault.cnf
P003-08-6-00.bin
P003-08-6-00.sbn
P0S3-06-3-00.bin
P0S3-06-3-00.sbn
P0S3-08-4-00.loads
P0S3-08-6-00.loads
P0S3-08-6-00.sb2
SEPMACADDRESS.cnf.xml
SIPMACADDRESS.cnf
(put your real MAC address in where it says MACADDRESS)
You can just manually enter the setting in the phone if you only have a few and did not want to setup a TFTP server and use the Cisco confg. files.
On the phone press the settings button (looks like a checkbox with a tick)
then go to Unlock Config the default password is cisco. Next go to the SIP configuration and Line 1 Settings and enter the details;
One Question. I think I am running firmware 8.2 in My phones. what are the benefits of 8.4, 8.6. Is 8.6 the latest? Where to get it on Cisco homepage is only 8.2 what I now of.
Also have anyone herd of any support for iax2 from Cisco phones?
Regards
Mattias