Erro Parsing SIPDefault.cnf Cisco 7960

arrundale · January 17, 2007, 12:17pm

I’m getting errors on my Cisco Phones 7960G (POS3-8-6-00) when booting up. The error is is:

W310 2 Error(s) Parsing: SIPDefault.cnf

Does anyone have any ideas why this would happen?

eisenmann · January 22, 2007, 1:12pm

Hello i have the same problem if i use the old FW 7.4 all is ok, but if i upgrade the FW to 8.6 i get also this error. In both i use the same config file.

JMG · April 25, 2008, 8:37pm

I’m having the same problem going from 7.7 to 8.6. Did either of you ever find anything out about the parsing error? Thanks, JMG

bwilks · April 26, 2008, 4:30am

Try changing the TOS to dscp (or just comment it out) in the SIPDefault.cnf It seem that Cisco changed it at some point.

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

#tos_media: 5
dscpForAudio: 184

telcoDave · April 27, 2008, 2:37am

I beat my brains out for 3 weekends straight with a similar issue. It wasn’t until I moved the the SEP…cnf.xml to a temporary directory so I could re-write the whole thing again that suddenly (while the phone continued to reset) the UAL recognized 8.8 through my SIPDefault.cnf and SIP.cnf.xml configs that it loaded and retrieved the settings in the SIP.cnf.xml file. YeeeHaaaa!!! I just issued the command ‘touch SEP.cnf.xml’ to create this file (the UAL looks for this file still even after the SIP FW load - at least that’s what ethereal is telling me) to keep it in my tftp directory. Everything looks OK except…

I can’t get my SIP’d 7960 or my SCCP’d 7921 to ring without “No such extension in context ‘default’” showing up in the CLI when I supposedly removed context default from sip.conf and extensions.conf files. Hmmm. Can’t get this one figured out.

Any Ideas???

telcoDave · April 27, 2008, 2:39am

I beat my brains out for 3 weekends straight with a similar issue. It wasn’t until I moved the the SEP…cnf.xml to a temporary directory so I could re-write the whole thing again that suddenly (while the phone continued to reset) the UAL recognized 8.8 through my SIPDefault.cnf and SIP.cnf.xml configs that it loaded and retrieved the settings in the SIP.cnf.xml file. YeeeHaaaa!!! I just issued the command ‘touch SEP.cnf.xml’ to create this file (the UAL looks for this file still even after the SIP FW load - at least that’s what ethereal is telling me) to keep it in my tftp directory. Everything looks OK except…

I can’t get my SIP’d 7960 or my SCCP’d 7921 to ring without “No such extension in context ‘default’” showing up in the CLI when I supposedly removed context default from sip.conf and extensions.conf files. Hmmm. Can’t get this one figured out.

Any Ideas???

bwilks · April 27, 2008, 3:40am

;Extensions.conf

[internal]

;Local Office Extensions
exten => _3XX,1,Dial(SIP/${EXTEN},45)
exten => _3XX,2,VoiceMail(${EXTEN})
exten => _3XX,3,Playback(vm-goodbye)
exten => _3XX,4,Hangup()

;sip.conf (extention 316) 

[316] 
disallow=all 
allow=ulaw 
allow=g729 
mailbox=316 
type=friend 
username=316 
callerid=316 
host=dynamic 
context=internal 
canreinvite=yes
secret=1234 
nat=no 
qualify=yes 
callgroup=1 
pickupgroup=1

;SIPDefault.cnf 

# SIP Default Generic Configuration File 

# Image Version 
image_version: P0S3-08-7-00 

# 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 

#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

vois · April 28, 2008, 6:37am

Hello i have the same problem if i use the old FW 7.4 all is ok, but if i upgrade the FW to 8.6 i get also this error. In both i use the same config file.

bwilks · April 28, 2008, 10:02am

Post your config

system · January 24, 2013, 7:15pm

Hi all,

Basically similar issue but I am trying to connect to sipgate.co.uk from home.
I am wondering if anyone can help me. I have CISCO 7960 which used to work with SIPgate.co.uk and stopped one day. I re-flashed it with latest SIP firmware but can not make it work now. My home router has UDP ports forwardings are set. Sipgate works from PCs, mobiles on the same network.

XMLDefault.cnf.xml
2000
2427
2428
P0S3-8-12-00
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
SipDefault.cnf
# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-8-12-00
# Proxy Server
proxy1_address: “sipgate.co.uk”	; Can be dotted IP or FQDN
proxy2_address: 222.33.22.11	 ; 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: 5069
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: 600
# 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: “”	 ; SNTP Server IP Address
sntp_mode: unicast	; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: GMT	 ; 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: 19	 ; Time of day in which DST starts
dst_stop_month: Jan	; Month in which DST stops
dst_stop_day: “”	 ; Day of month in which DST stops
dst_stop_day_of_week: Saturday	; 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: “sipgate.co.uk”	; Dotted IP of Backup Proxy
proxy_backup_port: 5060	; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: “sipgate.co.uk” ; 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: 1 ; 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: 1	; 0-Disabled (default), 1-Enabled
# Outbound Proxy Support
outbound_proxy: “proxy.live.sipgate.co.uk”	; 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 #######
# 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: 0	 ; 0-Disabled (default), 1-Enabled
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
SIP001380FDC397.cnf
image_version:P0S3-8-12-00
# Line 1 Setup
line1_name: “SipGate”
line1_authname: “34343434″
line1_shortname: “SipGate”
line1_password: “xxxxxx”
proxy1_address: “sipgate.co.uk”
proxy1_port: “5060″
line1_displayname: “VOIP Line 1″; # Line 1 Display Name (Display name to use for SIP messaging)
# Line 2 Setup
line2_name: “AnotherSip”
line2_authname: “Voip2″
line2_shortname: “Voip2″
line2_password: “Voip2″
line2_displayname: “VOIP Line 2″; # Line 2 Display Name (Display name to use for SIP messaging)
proxy2_address: “222.33.22.11″
proxy2_port: 5069
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: “My 7960″ ; Has no effect on SIP messaging
phone_prompt: “MySIP 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
telnet_level: 1

What is wrong with my configuration as non of the lines works. My priority is to fix Line 1 i.e. Sipgate at the moment but ideally to make both work.