Cisco 7940 Help

[kiwijonnyboy]

All,

I have been scouring the internet looking for configuration guidelines for the 79XX series Cisco IP phone. I have a 7940, and am trying to connect it to my voipo service.

I have loaded the SIP software and have successfully configured the TFTP server will all of the relavent files.

I have tried usernames with quotes\without quotes. I have also tried opening ports on the firewall with no success. However, all requests are driven from the SIP devices, so open ports should not be required. Am I missing something here?

I used wireshark to see what is happening and the phone is trying to register to the voipo server. It does this several times. I also get the INV to send to the voipo server when i dial a number. But the phone doesn't connect. All I get is the "Calling (out INV)" which is telling me it is sending the INV request.

I also used wireshark to see the registration from the voipo provided router and can see similar packet exchanges. I am thinking that I am missing a setting in my config but for the life of me can't find one that would matter on the registration.

I had X-lite running in 5 min. In the devices section, there is no Cisco phone, however X-lite shows up as well.

PLEASE HELP!

Below are my configurations:

SIPDefault.CNF

; sip default configuration file
#Image Version
image_version:P0S3-8-12-00 ;
#Proxy server address
proxy_register : 1
proxy1_address: "67.228.77.18"
proxy1_port : 5060
preferred_codec: g711ulaw

SIP<MAC>.CNF
phone_label: "lastname"
line1_name : voipophone#
line1_Shortname: "voipophone#"
line1_authname: "voipophone#"
line1_password: "BYODpass"
line1_displayname: "voipophone#"

[fisamo]

Does the phone not accept FQDNs instead of IP addresses? Try using sip.voipwelcome.com in the Proxy1_address field.

[kiwijonnyboy]

Just re-tried to verify and it still doesn't go. I can see from wireshark that it does resolve to the 67.xxx address.

Thanks for the quick reply.

Here are a couple lines from the IP Phone.
73 13.465243 192.168.0.107 67.228.77.18 SIP Request: REGISTER sip:sip.voipwelcome.com

265 51.764172 192.168.0.107 67.228.77.18 SIP/SDP Request: INVITE sip:3143212222@sip.voipwelcome.com, with session description


From my voipo router:
195 27.137418 192.168.0.109 174.37.45.134 SIP Request: NOTIFY sip:174.37.45.134

[kiwijonnyboy]

Figured out my issue.....

was missing the

nat_enable : 1

This was driving me NUTS!

This allows the external proxy to talk back to my IP address. Now I am trying to figure out the MWI. I think i may need to open some ports....but will save that for another day.

Once i get everything the way i like it....I will post up my configs.

[voipoh]

My 7960 config files. Don't know if it works for 7940

SIPDefault.cnf
# sip default configuration file
# Image Version for upgrade
;image_version: P0S3-8-12-00 ;
;image_version: P0S3-08-6-00 ;
;image_version: P0S3-07-5-00 ;
# Proxy server address
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

proxy1_address: "sip.voipwelcome.com" ; Can be dotted IP or FQDN
proxy2_address: "sipsorcery.com" ; Can be dotted IP or FQDN
proxy3_address: "orbtalk.co.uk" ; Can be dotted IP or FQDN
proxy4_address: "outgoing.future-nine.com" ; Can be dotted IP or FQDN
proxy5_address: "callcentric.com" ; Can be dotted IP or FQDN
proxy6_address: "proxy01.sipphone.com" ; Can be dotted IP or FQDN

# 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) ?changed to dscp
# tos_media: 5
dscpForAudio: 184
# 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
# 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: "194.81.227.227" ; SNTP Server IP Address (this is ntp1.ja.net)
sntp_server: "17.254.0.49" ; SNTP Server IP Address (this is ntp2.usno.navy.mil)
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EST ; 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: 0 ; 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: 2 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: "Nov" ; Month in which DST stops
dst_stop_day: 0 ; Day of month in which DST stops
dst_stop_day_of_week: "Sun" ; Day of week in which DST stops
dst_stop_week_of_month: 1 ; 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)
dnd_control: 0 ; Default 0 (Do Not Disturb feature is off)
callerid_blocking: 0 ; Default 0 (Disable sending all calls as anonymous)
anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls)
dtmf_avt_payload: 101 ; Default 101
# Sync value of the phone used for remote reset
sync: 1 ; Default 1
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)
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
nat_enable: 1 ; 0-Disabled (default), 1-Enabled
nat_address: 192.168.9.1 ; 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: "206.165.50.116" ; restricted to dotted IP or DNS A record only (this is fwdnat.pulver.com)
outbound_proxy_port: 5060 ; default is 5060
# 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: 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
# XML URLs
services_url: "http://phone-xml.berbee.com/menu.xml" ; URL for external Phone Services
# services_url: "http://www.ip-phone-services.com/bt/" ;bt services
# services_url: "http://your.site/services.xml" ; URL for external Phone Services
# services_url: "http://193.113.58.136/bt/" ;bt services

directory_url: "http://your.site/directory.xml" ; URL for external Directory location
logo_url: "http://hostsvg.com/logo.bmp" ; URL for branding logo to be used on phone display
# logo_url: "http://kermit/asterisk-tux.bmp"
# 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: 1 ; 0-Disabled (default), 1-Enabled
# Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
call_hold_ringback: 1 ; Default 0 (Disable ringback of held call)

SIPmacaddress.cnf
# phone-specific configuration file sample
line1_name : VOIPo
line1_authname : "xxx"
line1_password : "password"
line1_shortname : "Line 1"
line1_displayname : "VOIPo"
proxy1_address: "sipsorcery.com" ; Can be dotted IP or FQDN
proxy1_port: 5070

line2_name : Sipsorcery
line2_authname : "xxx"
line2_password : "password"
line2_shortname : "Line 2"
line2_displayname : "Sipsorcery"
proxy2_address: "sipsorcery.com" ; Can be dotted IP or FQDN
proxy2_port: 5080

line3_name: Orbtalk
line3_authname: "xxx"
line3_password: "password"
line3_shortname : "Orbtalk"
line3_displayname : "Line 3"
proxy3_address: "orbtalk.co.uk" ; Can be dotted IP or FQDN
proxy3_port: 5065

line4_name: F9
line4_authname: "xxx"
line4_password: "password"
line4_shortname : "Line 4"
line4_displayname : "F-Nine"
proxy4_address: "outgoing.future-nine.com" ; Can be dotted IP or FQDN
proxy4_port: 5062

line5_name: CallCentric
line5_authname: "xxx"
line5_password: "password"
line5_shortname : "Line 5"
line5_displayname : "CallCentric"
proxy5_address: "callcentric.com" ; Can be dotted IP or FQDN
proxy5_port: 5060

line6_name: Gizmo5
line6_authname: "xxx"
line6_password: "password"
line6_shortname : "Line 6"
line6_displayname : "Gizmo 5"
proxy6_address: "proxy01.sipphone.com" ; Can be dotted IP or FQDN
proxy6_port: 5061

####### New Parameters added in Release 2.0 #######
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "VoIP/Sorcery " ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "VOIPo"
line2_displayname: "Sipsorcery"
line3_displayname: "Ortalk"
line4_displayname: "F-Nine"
line5_displayname: "CallCentric"
line6_displayname: "Gizmo5"

####### New Parameters added in Release 3.0 ######
# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt: "SIP_7960: " ; 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

[voipoh]

Would appear that the main difference between 7940 and 7960 is 2 lines against 6
Some links.
http://www.markwilson.co.uk/blog/200...-using-sip.htm
http://www.mollien.net/index.php?mai...&article_id=10
http://www.voip-info.org/wiki/index....nts_style=flat

[man]

Figured out my issue.....

was missing the

nat_enable : 1

This was driving me NUTS!

This allows the external proxy to talk back to my IP address. Now I am trying to figure out the MWI. I think i may need to open some ports....but will save that for another day.

Once i get everything the way i like it....I will post up my configs.

Thank you for posting this, this is the exact same issue I had with my 7960.

I wish voipo support told me about this forum a month ago when I called for additional config info. I was only told I had what I need and others have no issues with the Cisco 7960 phone.

I was using my Cisco 7960 with Broadvoice and Gizmo5 without issues with the Nat off. I'm now able to connect to all three with the Nat On, BV and G5 doesn't seem to matter if NAT is on or off.

Thanks again kiwijonnyboy