What am I missing? I cant make or recive any calls on my 7960. softphones work perfectly. Below are the config files for the 7960. It does download them and shows the lines configured.
Many thanks
SIPDefault
Image Version
image_version: “P0S3-07-5-00”
Proxy Server
proxy1_address: "192.168.16.123"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: “”
Proxy Server Port (default - 5060)
proxy1_port:“5060"
proxy2_port:”“
proxy3_port:”“
proxy4_port:”“
proxy5_port:”“
proxy6_port:”"
Emergency Proxy info
proxy_emergency: "192.168.16.123"
proxy_emergency_port: “5060”
Backup Proxy info
proxy_backup: ""
proxy_backup_port: “5060”
Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: “5060”
NAT/Firewall Traversal
nat_enable: ""
nat_address: ""
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: “0”
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, g729)
preferred_codec: “none”
TOS bits in media stream [0-5] (Default - 5)
tos_media: “5”
Enable VAD (0-disable (default), 1-enable)
enable_vad: “0”
Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: “1” ; 0-Disabled, 1-Enabled (default)
Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: “0” ; 0-Disabled, 1-Enabled (default)
Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: “2” ; 0-Disabled (default), 1-Enabled, 2-Privileged
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: “0”
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 11
sip_invite_retx: “6” ; Default 7
timer_invite_expires: “180” ; Default 180 sec
Setting for Message speeddial to UOne box
messages_uri: “*97”
#********* Release 2 new config parameters **********
TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: “./”
Time Server
sntp_mode: "unicast"
sntp_server: "192.168.16.123"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: “1”
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-disabled, 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-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: “0” ; Default 0 (Disable blocking of anonymous calls)
Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: “1” ; Default 1 (Call Waiting enabled)
DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: “101” ; Default 100
XML file that specifies the dialplan desired
dial_template: “dialplan”
Network Media Type (auto, full100, full10, half100, half10)
network_media_type: “auto”
#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: “1”
#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: “0”
URL for external Phone Services
services_url: “http://192.168.16.123/xmlservices/index.php”
URL for external Directory location
directory_url: “http://192.168.16.123/xmlservices/PhoneDirectory.php”
URL for branding logo
logo_url: “http://192.168.16.123/cisco/bmp/trixbox.bmp”
SIP00036BDD3B32
Cisco SIP Configuration
phone_label: "cisco"
line1_name: "203"
line1_shortname: "203"
line1_displayname: "203"
line1_password: "203"
proxy1_address: "192.168.16.123"
proxy1_port: "5060"
line2_name: "UNPROVISIONED"
line2_shortname: "UNPROVISIONED"
line2_displayname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
line3_name: "UNPROVISIONED"
line3_shortname: "UNPROVISIONED"
line3_displayname: "UNPROVISIONED"
line3_password: "UNPROVISIONED"
line4_name: "UNPROVISIONED"
line4_shortname: "UNPROVISIONED"
line4_displayname: "UNPROVISIONED"
line4_password: "UNPROVISIONED"
line5_name: "UNPROVISIONED"
line5_shortname: "UNPROVISIONED"
line5_displayname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"
line6_name: "UNPROVISIONED"
line6_shortname: "UNPROVISIONED"
line6_displayname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"
line1_authname: "203"
line2_authname: "UNPROVISIONED"
line3_authname: "UNPROVISIONED"
line4_authname: "UNPROVISIONED"
line5_authname: "UNPROVISIONED"
line6_authname: “UNPROVISIONED”
asterisk1CLI> sip show peer 203
asterisk1CLI>
- Name : 203
Secret :
MD5Secret :
Context : from-internal
Subscr.Cont. :
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : “device” <203>
Expire : -1
Insecure : no
Nat : Always
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : (Unspecified) Port 0
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username:
SIP Options : (none)
Codecs : 0x40e (gsm|ulaw|alaw|ilbc)
Codec Order : (ulaw,alaw,gsm,ilbc)
Status : UNKNOWN
Useragent :
Reg. Contact :