[Asterisk-Users] Multiple lines on Cisco 7960

Scott Henderson scott at finite-tech.com
Fri Jan 7 10:56:40 MST 2005


I set this up manually on the phone and it works just fine so config 
files ...  I attached the complete config files so maybe someone can see 
what I am missing.

============
argon:/tftpboot# cat SIPDefault.cnf
# SIP Default Generic Configuration File
 
# Image Version
image_version: P0S3-07-3-00 ;

# Proxy Server
proxy1_address: "192.168.17.13"         ; Can be dotted IP or FQDN
proxy2_address: "192.168.17.13"         ; Can be dotted IP or FQDN
proxy3_address: "192.168.17.13"         ; Can be dotted IP or FQDN
proxy4_address: "192.168.17.13"         ; Can be dotted IP or FQDN
proxy5_address: "192.168.17.13"         ; Can be dotted IP or FQDN
proxy6_address: "192.168.17.13"         ; 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: none

# 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: "192.168.17.11"    ; SNTP Server IP Address
sntp_mode: directedbroadcast    ; unicast, multicast, anycast, or 
directedbroadcast (default)
time_zone: YST                  ; 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: "http://192.168.17.11/asterisk-tux.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-disabled, 1-enabled, 2-disabled no user control, 
3-enabled no user control)
call_hold_ringback: 0           ; Default 0 (Disable ringback of held call)

========================
argon:/tftpboot#  cat SIP00115C407FA3.cnf
# SIP Configuration Generic File
 
# Line 1
line1_name: Scott
line1_authname: "scott"
line1_password: "scott"

# Line 2
line2_name:  Scott1
line2_authname: "scott1"
line2_password: "scott1"

# Line 3
line2_name: "Line 2"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"

# Line 4
line2_name: "Line 4"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"

# Line 5
line2_name: "Line 5"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"

# Line 6
line2_name: "Line 6"
line2_authname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"

####### 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: "User ID"

# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: "User ID"


####### 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

messages_uri: "_6101"
argon:/tftpboot#

Nabeel Jafferali wrote:

>>I had not looked at the phones settings yet, thanks for the
>>suggestion. The setting indicate that there is no configuration on the
>>second line it is listed as "UNPROVISIONED"
>>    
>>
>
>Go into the phone and program Line 2 Settings directly, without using
>the SIP<MAC>.cnf file. If that works, then your .cnf file is wrong.
>
>  
>

-- 
Scott Henderson
============================================================================
Finite Technologies Incorporated
3763 Image Drive, Anchorage, Alaska 99504
Phone: 907.339.8085 ext 6101, Fax: 907.333.4482
http://www.finite-tech.com
http://www.chillywall.com
http://www.virtuale.cc
http://www.mphage.com
Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK
============================================================================

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