Freeswitch Cisco 7960G Howto

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This page was converted to confluence 4/29 by Mario G - DO NOT UPDATE




There will be more to come in this page, but I'm in the process of moving:


Firmware Revisions:

  • App Load: P0S3-08-8-00
  • Boot Load: PC030301
  • DSP Load: 4.0(2.0)[A0]

Also confirmed as working:

  • App Load: P0S3-08-11-00
  • Boot Load: PC030301
  • DSP Load: 4.0(5.0)[A0]

*Note: With previous revisions of the SIP Firmware for this phone it was discovered that NAT traversal was nearly impossible, and calls would mysteriously hang up after 120 seconds, etc.


Firmware Revision:

  • App Load: P0S3-08-11-00

This is confirmed as BUSTED! Weird things happen with this firmware revision, please consider using P0S3-08-8-00 until you're sure we're not kidding. Details: The bug in that version of Cisco firmware tends to create massive re-registration attempts, to the point that FS is so busy servicing the registration requests that it has no time to process incoming calls. When you have a few phones it will handle this without much of a problem, but when you add more you'll notice one-way audio and dropped call, adding yet more will create a situation whereby we've seen up to 25 seconds before an incoming call is even displayed on the CLI.

Really, no joke... Bad mojo... It took me a long time to figure this out, especially since with one phone everything in the test lab worked.

Update 11/19/2008 - This was a compounded problem having to do with a linux kernel bug, and it corrupting the DBD Database in Freeswitch. Oddly enough, this didn't surface until the aforementioned Cisco firmware weirdness above. This bug seems to effect linux, earlier and later versions may also be effected. The workaround was to put the Freeswitch SQLLite database in a RAM disk.



This HowTo assumes that you have a TFTP Server, have a DHCP Server configured to point the DHCP client devices at the IP address of the TFTP server, and that the Cisco 7960 Firmware listed above is installed properly on the TFTP server. There are many howto's that discuss how to install and/or upgrade the SIP Firmware Images on the TFTP server, so this is not covered in this HowTo.

For ISC DHCPD, you will need to have this line:

option tftp-server-name "your.tftp.server.ip";

in your dhcpd.conf

If that option doesn't work, or doesn't exist, try consulting your tftp server documentation for "option 150".

The Files:

You will need a SIPDefault.cnf file that sets up reasonable defaults for phones, and you will need one file created per phone with the MAC Address of the phone embedded into the filename such as SIP0A0B0C0D0E0F.cnf

The File Contents:

The contents of the SIPDefault.cnf file are listed below as it appears on my server. You may wish to update some items such as the sntp_server parameter and the date_format parameter depending on where you are in the world and/or the telnet_level depending on how secure you want things to be.

The auto_answer parameter is a bitmap, and as set will set line 6 to be capable of answering the call automatically. This can be useful for an intercom and paging for instance.


# SIP Default Generic Configuration File

# Image Version
image_version: P0S3-08-8-00

# 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: 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: 0

# 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: ""                ; 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: EST                  ; Time Zone Phone is in
dst_offset: 01/00               ; 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: Sunday   ; Day of week in which DST starts
dst_start_week_of_month: 8      ; Week of month in which DST starts
dst_start_time: 02/00           ; Time of day in which DST starts
dst_stop_month: October         ; Month in which DST stops
dst_stop_day: 0                 ; 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: 02/00            ; 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: M/D/Y

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 2                  ; 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: ""              ; 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 : 0             ; 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 #######

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: 1              ; 0-Disabled (default), 1-Enabled

####### New Parameters added in Release 4.4 #######

# Call Hold Ringback (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
call_hold_ringback: 3           ; Default 0 (Disable ringback of held call)

auto_answer: 32
autocomplete: 0
dnd_control: 2

The next file that needs to exist is the SIP<macaddress>.cnf file where <macaddress> is the hardware ethernet address of the phone itself. Hence: SIP00DEADBEEF00.cnf would pertain to a phone who's MAC Address is 00:DE:AD:BE:EF:00

You will need to replace host.domain.tld with the hostname or IP address of the SIP Profile you intend to use with the Cisco 7960's.

You will also need to replace the mysecret and 701X numbers with the credentials you created in the Freeswitch Directory. (more on that later).

The messages_uri is where you would specify the number to be dialed in order to retrieve voice mail. This corresponds directly to the number dialed by pressing the messages button, or the envelope button on the (7960 / 7960G).

**Be Advised: The demo dialplan that comes stock with freeswitch seems to expect the extensions to be 1XXX such as 1000, the configuration file below starts at 7010, and therefore you may need to make additional changes to the dialplan before you're able to call any of the provisioned lines on the phone.

SIP<mac address>.cnf

# SIP Configuration Generic File (start)

# Proxy Server
proxy1_address: "host.domain.tld"
proxy2_address: "host.domain.tld"
proxy3_address: "host.domain.tld"
proxy4_address: "host.domain.tld"
proxy5_address: "host.domain.tld"
proxy6_address: "host.domain.tld"

# Line 1 Settings
line1_name: "7010"                     ; Line 1 Extension\User ID
line1_displayname: "7010"           ; Line 1 Display Name
line1_authname: "7010"         ; Line 1 Registration Authentication
line1_password: "mysecret"         ; Line 1 Registration Password

# Line 2 Settings
line2_name: "7011"                          ; Line 2 Extension\User ID
line2_displayname: "7011"                   ; Line 2 Display Name
line2_authname: "7011"         ; Line 2 Registration Authentication
line2_password: "mysecret"         ; Line 2 Registration Password

# Line 3 Settings
line3_name: "7012"                          ; Line 3 Extension\User ID
line3_displayname: "7012"                   ; Line 3 Display Name
line3_authname: "7012"         ; Line 3 Registration Authentication
line3_password: "mysecret"         ; Line 3 Registration Password

# Line 4 Settings
line4_name: "7013"                          ; Line 3 Extension\User ID
line4_displayname: "7013"                   ; Line 3 Display Name
line4_authname: "7013"         ; Line 3 Registration Authentication
line4_password: "mysecret"         ; Line 3 Registration Password

# Line 5 Settings
line5_name: "7014"                          ; Line 3 Extension\User ID
line5_displayname: "7014"                   ; Line 3 Display Name
line5_authname: "7014"         ; Line 3 Registration Authentication
line5_password: "mysecret"         ; Line 3 Registration Password

# Line 6 Settings
line6_name: "7015"                          ; Line 3 Extension\User ID
line6_displayname: "7015"                   ; Line 3 Display Name
line6_authname: "7015"         ; Line 3 Registration Authentication
line6_password: "mysecret"         ; Line 3 Registration Password

# Emergency Proxy info
proxy_emergency: ""
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: "1"
nat_address: ""
voip_control_port: "5060"
start_media_port: "16384"
end_media_port:  "32766"
nat_received_processing: "0"

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "FreeSwitch"            ; Has no effect on SIP messaging

# Time Zone phone will reside in
time_zone: EST

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2"      ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Phone prompt/password for telnet/console session
phone_prompt: "C7960"                              ; Telnet/Console Prompt
phone_password: "myphonesecret"                          ; Telnet/Console Password

# Enable_VAD (1-enabled, 0-disabled)
enable_vad: "0"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"
user_info: phone

# URL for external Directory location
#logo_url: ""                    ; URL for branding logo to be used on phone display

# SIP Configuration Generic File (stop)

# Insert Extension for voice mail retrieval here, this is what's dialed when they hit the "envelope" key on the phone.
messages_uri:  ""

# This is a bitmap, and it's supposed to set line 6 for auto-answer.  
#However it seems that the newer versions won't allow anything above line4 to be set to auto-answer via tftp.
auto_answer: 32
autocomplete: 0
dnd_control: 2

BLIND XFER (Blind Transfer):

While the attended transfer works. the blind transfer exhibited a problem when a call was placed to the telephone as an intercom call where the line was automatically answered.

In my defaut dialplan context I had to add this right at the top in order for the device to properly transfer calls that were placed to it's intercom line:

<extension name="Cisco Xfer Fix">
      <condition field="destination_number" expression="^(auto-answer)$">
	<action application="set" data="destination_number=${sip_to_user}"/>

Call Parking

There doesn't seem to be a way to provision the bottom keys on the phone, so parking has been implemented as a blind transfer to a particular extension in my dial plan. Frankly I'm disappointed, but hey it's Cisco.

Conferencing and Intercom

The Cisco 7960 and/or 7960G don't implement a multicast paging group, so there is no paging and intercom. See above link on how to implement it with unicast and a conference bridge.

More to come after I move...