FW: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Matt Schulte
mschulte at netlogic.net
Thu Dec 16 10:05:21 MST 2004
ala cisco 7960
-----Original Message-----
From: Matt Schulte
Sent: Thursday, December 16, 2004 10:34 AM
To: 'Paul A Brown'
Subject: RE: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Sure thing, the biggest problem I had was getting the SIP filenames
working correctly for updating the firmware (blah, I love Cisco but
these phones are a joke for support). This works for me! Good luck.
sip.conf
[107]
host=dynamic
type=friend
context=default
username=107
secret=blahblah
mailbox=107
canreinvite=no
disallow=all
allow=all
--------------------------
-sipMACADDRESS.cnf-
image_version: P0S3-07-3-00
line1_name: 107
# Line 1 Registration Authentication
line1_authname: "107"
# Line 1 Registration Password
line1_password: "elblahblah"
--snip--
####### 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: "Matt S 107" ; Has no effect on SIP messaging
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Matt S"
# 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: "blahblahblah" ; Limited to 31 characters (Default -
cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
-------------------------
sipdefault.cnf
# Image Version
image_version: "P0S3-07-3-00"
# Proxy Server
# Note: I put the proxy server information in the individual conf files
# for each machine, since each box has different configs. You could, of
course, # put all of them here in the Default file...
proxy1_address: "192.168.1.17"
#proxy2_address: "192.168.117.4"
# Proxy Server Port (default - 5061)
#proxy1_port:"5060"
# Emergency Proxy info
proxy_emergency: "192.168.1.17"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "192.168.1.17"
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: "192.168.1.17"
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "0"
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: "120"
# 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: "1" ; 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: "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 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec
# Setting for Message speeddial to UOne box
messages_uri: "8500"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "17.254.0.49"
time_zone: "CST"
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: "1"
####### New Parameters added in Release 4.0 #######
# XML URLs
services_url: "http://who.cares/blah/" ; test url
directory_url: "http://192.168.1.17/directories.xml"
# URL for external Directory location
logo_url: "http://192.168.1.17/n2voiplogo.bmp" ; URL
for branding logo to be used on phone display
# put your own logo in the logo_url location; I include the 10-20.com
one for reference in building your own
# HTTP Proxy Support
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)
# The dynamic tftp server should be set to whatever your TFTP server is.
This way, it # keeps the tftp server setting even though you might be
using DHCP (default behavior # is to use the DHCP server as a tftp
server, which is rarely correct.)
# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
-----Original Message-----
From: Paul A Brown [mailto:paul at fowlmere.com]
Sent: Thursday, December 16, 2004 10:07 AM
To: Matt Schulte
Subject: Re: [Asterisk-Users] Cisco 7960 (SIP) hold problems
Hi Matt,
Seems you have gone one step further than me,
Can you possibly let em see your sip configs for the cisco 7960 and also
the
configs in the phone. I am having trouble getting it to talk to asterisk
Thanks
Paul
----- Original Message -----
From: "Matt Schulte" <mschulte at netlogic.net>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, December 16, 2004 3:01 PM
Subject: [Asterisk-Users] Cisco 7960 (SIP) hold problems
> Has anyone had problems with using hold on a 7960 SIP firmware? The
> problem is when the 7960 puts a call on hold and you take it off hold
> again, the 7960 outbound audio is delayed on the other end. Sometimes
> up to a few seconds. I've tried a couple different things, making the
> "other end" a diff type of trunk ie:
>
> 7960sip --> asterisk --> IAX2 --> PRI
>
> 7960sip --> asterisk --> SER --> SIP proxy
>
> Anyone have a clue? The 7960 has the latest firmware, 7.3 or
> something. Could this be a (the?) problem? Thanks!
>
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