Cisco 7940 connection problems

hello friends i have a bit of a crazy question and im hoping i can get a few leads on how to fix this.

i have 2 Cisco 7940 phones that currently run SIP (i know this by the SIP logo in the top right corner) they are both using and successfully load firmware P0S3-8-12-00 as i get no error messages on my phones status menu other than my time server, backup and emergency not being configured currently.

i have an HP computer which is running a freePBX version 12.0.43 connected to a Cisco 3750 switch that has PoE and the phones are also connected to the switch (This is not connected to the outside world). i have also installed on my freePBX machine webmin version 1.730 and the DHCP module configuring my networking for 192.168.3.X network (this is just for simulation purposes so i can better understand VoIP). the DHCP server runs and issues ip address to clients and does not cause any of my freePBX services to crash which is good. but i can’t call one phone from the other. when i pick up the phone with extension 101 and dial extension 102 and hit dial it says Calling (out INV) for about 63 seconds and then does the disconnected beep beep beep. does anyone know what im doing wrong or what that Calling(out INV) means. im pretty computer literate and am majoring in CIS so if you have any questions about my question im all ears!

i have configured my SIPDefault.cnf file as follows

SIP Default Generic Configuration File

Image Version

image_version: P0S3-8-12-00

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 o
proxy6_address: “” ;

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: “” ; 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: “76.XXX.XXX.XXX” ; <== my actuall ip address
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: 10 ; 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 #######
; 0-Disabled (default), 1-Enabled


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

#Voice Mail extention
messages_uri: 8500

RTP Call Statistics (SIP BYE/200 OK message exchange)

call_stats: 0

#Transfer by hanging up the phone

i believe this to be the correct configuration but please share your knowledge.

i have configured my SIP.cnf files all the same way (other than the mac address in the file name and the values change based on each phone but the formatting is the same) i’ll provide my example below

Line 1 Registration Information

line1_name: “101” ; Extension number
line1_authname: “101” ; Extension number
line1_shortname: “kevin” ; Name that appears on t
line1_password: “cisco123” ; Seceret Password

Line 2 Registration Information

line2_name: “” ; Extension number
lne2_authname: “” ; Extension number
line2_shortname: “” ; Name that appears on the screen, please set this line to autoanswer
line2_password: “” ; Seceret Password

Line 1 Display Name (Display name to use for SIP messaging)

line1_displayname: “kevin” ;
line2_displayname: “” ;

Label shown in upper right corner just before the stylized “sip”

phone_label: "Phone loaded mak .cnf! "

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

so the only thing i have left is to do a little configuring on the phone right?

i go to configure the Sip Line settings on my phones by going Setting->SIP configuration->Line 1 and i have the configuration as follows in relation to the SIP.cnf file i posted above.
Name : 101
Shortname: kevin
Authentication Name: 101
Authentication Password: cisco123
Display Name: kevin
Proxy Address:
Proxy Port: 5060

my general SIP configurations on all my phones are as follows
Messages url: 8500
Preferred Codec: g711ulaw
Out of Band DTMF: avt
Register with Proxy: yes
Register Expires: 3600
TFTP Directory: (NULL)
Phone Label : Phone loaded mak.cnf!
Enable VAD: no
VoIP Control Port: 5060
Start Media Port: 16384
End Media Port: 32766
backup Proxy: (NULL)
Backup Proxy Port: 5060
Emergency Proxy: (NULL)
Emergency Proxy Port: 5060
Outbound proxy
Outbound proxy port: 5060
NAT Enabled: no
NAT Address: 76.XXX.XXX.XXX

i know freepbx recognizes my extensions because it says they are offline, my sip extension settings in freepbx are as follows

Extension 101
Phone Type: CHAN_SIP
Display Name: kevin
Secret: cisco123
Connection Type: friend
Nat: NO

did i miss something?

I could be talking out of my a$$ here but I have seen those 7940/60s do some wierd things with ignoring port number settings. I suspect your issue has something to do with webmin listening on port 10000 by default. I had that tank a 7960 on me once. Stopped webmin service and the problem went away. I know the setting below should work but I have personal experience with the settings below being ignored by these phones.

VoIP Control Port: 5060
Start Media Port: 16384
End Media Port: 32766

I was trying to use 5061 at the time and after about 10 hours of sip debugging I could not get the phone to use anything but what they thought they should be using port wise. If just shutting down webmin doesn’t work then I would look at using the default ports on everything on the phones and the pbx. Also I would cut out about 90% of those settings for further testing.

i know this may be a loaded question but what kind of things can i remove from my SIPDefault.cnf file? i appreciate the help though i will try that once i get back from my database class and let you know what happens

This is my SIPDefault.cnf

image_version: P0S3-07-5-00
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_register: 1
messages_uri:   "1"
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
;sntp_mode: unicast
;sntp_server: ""
;time_zone: "GMT" ; assuming you're in GMT
time_format_24hr: 1 ; to show the time in 24hour format
date_format: "Y/M/D"  ; format you would like the date in
dial_template: dialplan

This is a config from a phone(SIPmac.cnf). Even this one has more then the bare minimum. I would start here though.

image_version: P0S3-07-5-00
line1_name: 404
line1_authname: "404"
line1_shortname: "Line 1" ; displayed on the phones softkey
line1_password: "XXXXXXXXX"
line1_displayname: "Busty"; the caller id
proxy1_port: 5060
# Line 2 Setup
line2_name: 404
line2_authname: "404"
line2_shortname: "Line 2"
line2_password: "XXXXXXXXXX"
line2_displayname: "Busty";

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Some Company, Inc. x404 " ; add a space at the end, looks neater
phone_password: "cisco" ; Limited to 31 characters (Default - cisco)
user_info: none
telnet_level: 2
messages_uri: *97

I would also default the media ports on your freepbx box for testing. Like I said I have seen these 79xx phones ignore port directives in configs. You watch the sip debug on the phone it is pretty much doing what ever it wants to. Get it to register with the defaults and bare minimum and go from there. Also there is a bug with anything newer then I believe 7-5 where the phone won’t reregister if it loses it’s connection for a while(worry about that later). I have seen it personally and read about it on the net which is why I am using 7-5. All the firmware versions are free from cisco below 8. You can get 8 with a smartnet which I have(it was like $30 but a waste cause it has bugs). I have a ton of these phones. Some in other countries even. They all work flawlessly. If you need further help I might be more inclined to have a phone conversation because I am slammed these days and don’t have a lot of time for typing stuff out. Are you in the US?

Another thought. It seems to me your problem is that the phone is not registering. Did you telnet to the phone and check it registration status?