Cisco 7960 - bekomme es einfach nicht hin

[buedims]

Hallo zusammen,
ich setze seit einiger Zeit ein Grandstream GXP2000 problemlos ein. Nun habe ich von meiner Firma ein nicht mehr benötigtes 7960 bekommen, dass vorher per SCCP genutzt wurde. Ich habe mir dann die neueste SIP Firmware 8.2 von Cisco heruntergeladen und mit meinem Sipgate Account konfiguriert. Auf meinem DSL Router habe ich Port 5060 weitergeleitet und auch mal Testweise das Phone als DMZ Host gekennzeichnet, aber es will einfach nicht richtig funktionieren. Direkt nachdem ich das Phone gebootet habe, kann ich raustelefonieren, aber es kann mich keiner erreichen. Wenn dann einige Zeit vergangen ist, kann ich weder telefonieren noch angerufen werden. Ich nehme an, dass ich in der SIPDefault.cnf einige Fehler habe. Da aber die ganze Cisco Seite völlig neu für mich ist, wollte ich mal nachfragen, ob hier jemand ist, der mir etwas auf die Sprünge helfen kann. Hier ist meine CNF Daten.

# SIP Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: ""		; 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: 500

# 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: "ntp.sipgate.net"			; SNTP Server IP Address
sntp_mode: directedbroadcast	             ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET			       ; 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)
date_format: D-M-YY		; Dateformat Day, month, year

# 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: 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: 5004	 	; Start RTP range for media (default - 16384)
end_media_port: 5007	   	; End RTP range for media (default - 32766)
nat_received_processing: 1	; 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: "http://www.fo-pa.de/cgi-bin/rss2cisco.pl"		; URL for external Phone Services
directory_url: "//niels/tftp-root/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 (Message Waiting Indicator)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
stutter_msg_waiting: 1          ; 0-Disabled (default), 1-Enabled

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

# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
autocomplete: 0                 ; 0-Disabled, 1-Enabled (default)

 

[chaos2000]

hallo buedims,

du hast in der conf doch gar keine proxy1_address definiert, ist das richtig?
Es ist auch zu empfehlen die nat_address zu setzten - z.B. mit einem dyndns-Account;
Welchen Anbieter verwendest du?

 

[CTU]

@Chaos und buedims :

Ist der Bereich 5004-5007 für die Mediaports überhaupt richtig.

Ich habe da was mit 16000 - 17000.





mfg,

CTU

 

[buedims]

@chaos2000

wenn ich als nat_address meine dyndns Adresse eingebe, bekomme ich garkeinen Connect mehr. Mein DSL Provider ist Arcor. Mein SIP Account habe ich bei Sipgate.

Wenn ich die dyndns Adresse weglasse, bekomme ich zwar einen Connect zu Sipgate, aber es funktioniert wie oben beschrieben nur direkt nach dem booten und anrufen kann man mich garnicht. Ich kann wenn überhaupt nur rauswählen.

Was muss ich denn bei proxy1_address hinschreiben?

 

[chaos2000]

proxy1_address ist die adresse von sipgate

@CTU

ja, das ist ok; bei sipgate wird auch dieser Bereich als Beispiel genommen

 

[buedims]

Hallo,

habe prei proxy_1 address jetzt "sipgate.de" eingetragen, aber es ist erfolglos. Sobald ich unter "nat_address" meine dyndns Adresse eintrage, bekomme ich keine Verbindung mehr zu Sipgate (telefon mit kleinem "x" auf dem Display). Nehme ich die nat_address weg habe ich zwar Verbindung zu Sipgate, habe aber dann das Verhalten, wie ich es ganz oben beschrieben habe. Ich weiss nicht mehr weiter.

Hat irgendjemand vielleicht noch eine Idee?

 

[cobra]

koenntest Du dann mal den aktuellen Stand Deiner SIPdefault.cnf und SIPMac.cnf posten.

 

[chaos2000]

ich würde auch gern noch einmal beide dateien anschauen, die passwörter aber unkenntlich machen

 

[buedims]

OK, hier ist meine sipmac.cnf

# SIP Configuration Generic File

phone_label: "CISCO 7960"

######################################### Line 1 - Sipgate

proxy1_address: "sipgate.de"
proxy1_port: 5060
line1_name: "xxxx"
line1_authname: "xxxx"
line1_password: "xxxx"
line1_displayname: "Sipgate"
line1_shortname: "SipGate"


######################################### Line 2 - Sipgate

proxy2_address: "nokitel.de"
proxy2_port: 5060
line2_name: ""
line2_authname: ""
line2_password: ""
line2_displayname: ""
line2_shortname: "nokitel"

######################################### Line 3 - web.de

proxy3_address: "sip.web.de"
proxy3_port: 5060
line3_name: ""
line3_authname: ""
line3_password: ""
line3_displayname: ""
line3_shortname: "Web.de"

######################################### Line 4 - Sipsnip

proxy4_address: "sipsnip.com"
proxy4_port: 5060
line4_name: ""
line4_authname: ""
line4_password: ""
line4_displayname: ""
line4_shortname: "Sipsnip"

######################################### Line 5 - Purtel

proxy5_address: "deu1.purtel.com"
proxy5_port: 5060
line5_name: ""
line5_authname: ""
line5_password: ""
line5_displayname: ""
line5_shortname: ""

######################################### Line 6 - Stanaphone

proxy6_address: "proxy01.sipphone.com"
proxy6_port: 5060
line6_name: ""
line6_authname: ""
line6_password: ""
line6_displayname: ""
line6_shortname: "Sipphone"

####### 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: "Area72  "	; 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: ""

####### 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

Und hier ist die Sipdefault.cnf

# SIP Default Generic Configuration File
########################################
# Image Version
# Je nachdem welche Image Version Verwendung findet,
# den Eintrag entsprechend abaendern...
image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: "sipgate.de"		; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060

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

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 500

# 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: "ntp.sipgate.net"			; SNTP Server IP Address
sntp_mode: directedbroadcast	             ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET			       ; 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)
date_format: D-M-YY		; Dateformat Day, month, year

# 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: 1                 ; 0-Disabled (default), 1-Enabled
nat_address: "xxxx.dyndns.org"    		      ; 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: 16390	   	; End RTP range for media (default - 32766)
nat_received_processing: 1	; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: "sipgate.de"	 	; 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: "http://www.fo-pa.de/cgi-bin/rss2cisco.pl"		; URL for external Phone Services
directory_url: "//niels/tftp-root/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 (Message Waiting Indicator)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
stutter_msg_waiting: 1          ; 0-Disabled (default), 1-Enabled

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

# Telefonnummern automatisch vervollstaendigen (macht bei mir Probleme, also aus)
#**** 0-abgeschaltet
#**** 1-eingeschaltet
autocomplete: 0                 ; 0-Disabled, 1-Enabled (default)

 

[CTU]

Also was ich dir spontan empfehelen würde , egal ob es was bewirkt oder nicht , doppelte einträge zu vermeiden. Z.b. sehe ich , dass Proxy_Port und Proxy_Adress doppelt vorkommt. Einmal reicht es , also in einer der beiden Dateien weglassen.






CTU

 

[cobra]

Also so auf Anhieb faellt mir nur auf, dass bei line1_name keine Anfuehrungszeichen sein sollten, also

line1_name: xxxx
line1_authname: "xxxx"

PS. Wer oder was ist Nokitel?

 

[buedims]

Hallo,

habe alles ausprobiert, aber es will einfach nicht klappen. Vielleicht liegt es ja auch am DSL Router oder an meinem DNS Server. Ich glaube ich gebe auf. Mein Grandstream GXP 2000 funktioniert ja einwandfrei.

Trotzdem vielen Dank für Eure Hilfe.

Gruß Buedi

 

[chaos2000]

@buedims,

so schnell schon aufgeben ?

ich habe noch mal meine SIPDefault angeschaut, da habe ich 2 Unterschiede:

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

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

Was mir noch dazu einfällt ist, dass

line1_name und line1_authname gleich sein müssen (geht auch mit "").
Wenn Du willst kann ich mal meine cnf schicken - als Bsp

 

[buedims]

@chaso2000

Danke, das wäre nochmal ein Versuch wert. Poste doch hier einfach Deine CNF Files, die ändere ich dann mit meinen Daten ab und schiebe sie mal auf das Cisco. Vielleicht klappt das ja.

Buedi

 

[chaos2000]

ok, hier noch mal meine files


SIPDefault.cnf

# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-08-2-00

# Proxy Server
proxy1_address: "sipgate.de"		; Can be dotted IP or FQDN
proxy2_address: ""			; 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_always

# 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: "./cisco_voipphones/"		; Example:  ./sip_phone/

# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
#sntp_server: "217.10.79.4"			; SNTP Server IP Address
sntp_server: "192.53.103.103"			; SNTP Server IP Address ptbtime1.ptb.de
;sntp_mode: directedbroadcast	; unicast, multicast, anycast, or directedbroadcast (default)
sntp_mode: unicast		; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CET			; Time Zone Phone is in
dst_offset: 1			; Offset from Phone's time when DST is in effect
dst_start_month: March		; Month in which DST starts
dst_start_day: ""		; Day of month in which DST starts
dst_start_day_of_week: Sunday	; Day of week in which DST starts
dst_start_week_of_month: 8	; 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: 1			; 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: "UNPROVISIONED"		; Dotted IP of Backup Proxy
proxy_backup_port: 5060		; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "UNPROVISIONED" 		; 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: stun.sipgate.net:10000    ; 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: 30000 	; Start RTP range for media (default - 16384)
end_media_port: 30010   	; End RTP range for media (default - 32766)
nat_received_processing: 1	; 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: 2			; 0-Disabled (default), 1-Enabled, 2-Privileged

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

# XML URLs
services_url: http://www.voip-info.de/feed/cisco7960/index.xml		; URL for News-feed


directory_url: ""		; URL for external Directory location

logo_url: "http://asterisk.local.net/voip/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-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: 1		; 0-Disabled (default), 1-Enabled

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

SIP<MAC>.cnf

# SIP Configuration Generic File 
 
# Line 1 appearance
line1_name: "SIPGATEID"

# Line 1 Registration Authentication 
line1_authname: "SIPGATEID"

# Line 1 Registration Password
line1_password: "password"

# 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: "Testphone"	; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "sipgate"

# 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: "cisco" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none