atcomsystems.ca/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems 8002 help!!!

Have a CIX100 with a MIPU and current firmware.

Put 3CDaemon TFTP on the MicroMAS-H, got the file from Toshiba, modified the sip_user.cfg file for ext 219. Created ext 219 as SIP in the 100 and changed the protocol.

I'm using a Ubiquity Picostation2 that supports WMM.

The handset will seemingly at random work. It will SOMETIMES connect and pull the config file down no problem, other times the TFTP server says the "file can't be read".
When it does connect and I can dial out it works beautifully but typically won't work for more than 3 times in a row, sometimes even the second call will result in "rejected" or "busy".

Inbound has only worked once. Most of the time I'll dial the extension 219 and the digital phone will sit there like I haven't dialed enough digits. Sometimes it'll show Vacant.

I'm completely baffled as to how it can work so sporadically! Toshiba is virtually useless in troubleshooting and Polycom acts like it isn't their problem either! I could sell a bunch of these handsets if I could just get them to work!!!

Will need to know the IP Address of the IPU/ handset and the MAS IP Address.
We will also need to see your .cfg files "sip_user.cfg/ allusers.cfg.
You also need to tell us how the handset is programmed.

Tech support should be looking at the following;
The CIX is programmed properly (IPU/Station/License) May even ask about the netwrk, depends on who you get you may get the full service with review of your .cfg files

Handset IP is 192.168.168.219
MIPU 192.168.168.251
MAS 192.168.168.252

SIP_219.cfg
# Codec preference order only. This does not enable/disable codecs.
# (Optional) can be G.711-ulaw, g.711u, G.711U, g711u, G.711U, etc.
# if g711u is omitted it will be added to end of list.
# if g711a is omitted it will be added to end of list after u.
CODECS = g711u, g711a
# One PROXYn (PBX/Call Server) is required, additional ones are optional as
# you can register secondary line appearances with other PROXY servers
#
PROXY1_TYPE = TOSHIBA
PROXY1_ADDR = 192.168.168.251:5060
#PROXY2_ADDR =
# ProxyDomain can be omitted if a specific proxy domain name is not defined
# at the proxy server. If omitted, the ProxyDomain defaults to the IP address
# of the proxy server.
# (below are examples of different ways to specify a domain)
#PROXY1_DOMAIN =
#PROXY1_DOMAIN =
#PROXY1_DOMAIN =
# PROXY1_MAIL_SUBSCR is who we should subscribe to for mail center
# notifications
# This is needed only if the user is not subscribed automatically at
# registration.
# It is almost never required in current versions of Asterisk to specify
# this.
# If you are using Asterisk (non-business edition) before v1.2, this is
# necessary.
# This example is actually specific for a line number:2425
#PROXY1_MAIL_SUBSCR = sip:[email protected]
# PROXYn_MAIL_NOTIFY is from whom we might get unsolicited mail center
# notifications
# This option is deprecated and no longer needed in versions beyond
# 8002/e340/h340/i640 handsets v108.011, Polycom 8002 phones v130.001, and
# Polycom 8020/8030 phones 131.001.
# Examples:
#PROXY1_MAIL_NOTIFY =
#PROXY1_MAIL_NOTIFY =
# PROXY1_MAIL_ACCESS is the main voicemail dial number
# Examples:
# PROXY1_MAIL_ACCESS =
# PROXY1_MAIL_ACCESS =
# PROXY1_MAIL_ACCESS =
#PROXY1_KEYPRESS_2833 controls generation of in-stream RFC2833 formatted key
# press events. Normally you want this to be disabled for Asterisk but it
# depends on your configuration and what you want to be able to do.
# The default is disable
PROXY1_KEYPRESS_2833 = enable
# PROXY1_KEYPRESS_INFO controls generation of SIP INFO requests to the SIP
# server for keypress events. Normally you want this to be enabled.
# The default is enable
PROXY1_KEYPRESS_INFO = disable
# PROXYn_HOLD_IP0 controls setting of media stream IP destination to 0.0.0.0
# when a call is put on hold.
# PROXYn_HOLD_IP0 is not required for current versions of Asterisk.
# For older PBXs that require this, set this to enable
PROXYn_HOLD_IP0 = enable
# PROXY1_PRACK enables ACK'd provisional responses to INVITE requests. The
# PRACK mechanism will be used if this switch is enabled and the Proxy server
# specifies support for the PRACK mechanism. PRACK is NOT SUPPORTED in
# current versions of Asterisk, but is to be supported on subsequent
# versions. PRACK should not be required on local area networks.
PROXY1_PRACK = enable
# ////////////////////////
# ////// ABOVE this line should probably be in the sip_allusers.cfg file
# ////// with items uncommented in this file only for overriding a setting
# ////// for a particular user.
# ////////////////////////
# Authentication credentials
# (Normally not stored in this file for security reasons)
# AUTH = username; password
AUTH = 219;
# Line definitions
# Each definition should have LINEn, LINEn_PROXY and LINEn_CALLID
# LINEn is the dial number
# LINEn_PROXY is the PROXYn server this line should register with, typically
# defined in sip_allusers.cfg.
# LINEn_CALLID is shown on the standby display of 8020/8030 phones but not
# 8002 or e/h340/i640 phones. The Asterisk Server converts the callID
# information to alternative forms defined in the Asterisk configuration
# files for display at the far end of a phone call.
# Up to 5 line definitions can be made for each user
# Line definitions do not necessarily have to have different extensions
LINE1 = 219
LINE1_PROXY = 1
LINE1_CALLID = Wireless
# Two lines may map to the same extension to allow second incomming calls.
#LINE2 = 3739
#LINE2_PROXY = 1
#LINE2_CALLID = Brady, Marsha
#LINE3 = 3739
#LINE3_PROXY = 1
#LINE3_CALLID = Drew, Nancy
#LINE4 =
#LINE4_PROXY = 3
#LINE4_CALLID = Sip User 4
#LINE5 =
#LINE5_PROXY = 2
#LINE5_CALLID = Sip User 5
# Favorite Dialed Number list.
# You can define up to 8 total entries including any defined in
# sip_allusers.cfg.
# You can enclose a string in quotes to allow for spaces.
# Each favorite can be complete SIP URI
# Format is:
# FAVORITE = dial_string; username
#
# The username can be blank and can include escaped chars.
#FAVORITE = "201"; "Bob"
#FAVORITE = 202; "Jill in Accounting"
#FAVORITE = 3600; "SoundPoint 3013"
#FAVORITE = 203; "Jane"
#FAVORITE = "9495833600"; "Richard's Cell"

sip_allusers.cfg

# SIP ALL USERS Configuration file example

# Configuration file format example

# Codec preference order only. This does not enable/disable codecs. (Optional)
# can be G.711-ulaw, g.711u, G.711U, g711u, G.711U, etc.
# if g711u is omitted it will be added to end of list.
# if g711a is omitted it will be added to end of list after u.
CODECS = g711u, g711a
PROXY1_TYPE = TOSHIBA
# One PROXYn_ADDR (PBX/Call Server) is required, additional ones are optional as you
# can register secondary line appearances with other PROXY servers
#
PROXY1_ADDR = 192.168.168.251:5060
#PROXY2_ADDR = 172.29.0.140:5060

#ProxyDomain can be omitted if a specific proxy domain name is not defined at the
# proxy server. If omitted, the ProxyDomain defaults to the IP address of the
# proxy server.
# (below are examples of different ways to specify a domain)
#PROXY1_DOMAIN = plcmengr.com
#PROXY1_DOMAIN = 10.0.0.138
#PROXY1_DOMAIN = axlx.engr.local

# PROXYn_MAIL_SUBSCR is who we should subscribe to for mail center notifications
# This is needed only if the user is not subscribed automatically at registration.
# It is almost never required in current versions of Asterisk to specify this.
# If you are using Asterisk (non-business edition) before v1.2, this is necessary.
# This example is actually specific for a line number:3001
#PROXY1_MAIL_SUBSCR = sip:[email protected]


# PROXYn_MAIL_NOTIFY is from whom we might get unsolicited mail center notifications
# This option is deprecated and no longer needed in versions beyond e/h340/i640
# phones v108.011, Polycom 8002 phones v130.001, and Polycom 8020/8030 phones
# 131.001.
# Examples:
#PROXY1_MAIL_NOTIFY = [email protected]
#PROXY1_MAIL_NOTIFY = sip:[email protected]

# PROXYn_MAIL_ACCESS is the main voicemail dial number
# Examples:
# PROXY1_MAIL_ACCESS = 7999
# PROXY1_MAIL_ACCESS = sip:[email protected]
# PROXY1_MAIL_ACCESS = [email protected]
#PROXY1_MAIL_ACCESS = 7999

#PROXYn_KEYPRESS_2833 controls generation of in-stream RFC2833 formatted key
# press events. Normally you want this to be disabled for Asterisk but it depends
# on your configuration and what you want to be able to do.
# If you are going to do OAI integration, this must be disabled.
# The default is disable
PROXY1_KEYPRESS_2833 = enable

#PROXYn_KEYPRESS_INFO controls generation of SIP INFO requests to the SIP server
# for keypress events. Normally you want this to be enabled.
# If you are going to do OAI integration, this must be enabled.
# The default is enable
PROXY1_KEYPRESS_INFO = disable

# PROXYn_HOLD_IP0 controls setting of media stream IP destination to 0.0.0.0 when
# a call is put on hold.
# PROXYn_HOLD_IP0 is not required for current versions of Asterisk.
# For older PBXs that require this, set this to enable
PROXYn_HOLD_IP0 = enable

# PROXYn_PRACK enables ACK'd provisional responses to INVITE requests. The PRACK
# mechanism will be used if this switch is enabled and the Proxy server specifies
# support for the PRACK mechanism. PRACK is NOT SUPPORTED in current versions of
# Asterisk, but is to be supported on subsequent versions. PRACK should not be
# required on local area networks.
PROXY1_PRACK = enable

# PROXYn_KEEPALIVE_SECS specifies that the phone should send Keep alives to the
# PROXYn server. It can be set to 0 or anywhere from 10 to 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 0 which means the phone will not send keep_alives to PROXYn.
#
# If the phone failes to get a keepalive response within the SIP 32 second
# timeout, the keepalives are terminated for that PROXY until the next successful
# registration to the PROXY.
#
# This parameter is REQUIRED on Asterisk Business Edition with the Polycom 8002
# phone but not with the Polycom 8020/8030 phones. This is because Linux, the
# OS running Asterisk, has a ARP Cache flushing mechanism. This feature serves
# to refresh the ARP Cache for the 8002 phones. For 8020 and 8030 phones, the
# ARP Cache is refreshed by the SVP server. 8020/8030 phones do not know how to
# parse this parameter.
#
# The value of this parameter for 8002 phones should be just less than the
# minimum ARP Cache flushing time on the Linux box running Asterisk. For Redhat
# this minimum flushing time is 15 seconds, so the parameter needs to be set for
# 14 seconds or less. For other Linux Distributions, you may have to determine
# what this minimum value is, however 14 seconds should work fine.
#
# The lower this setting, the more the impact on the 8002's standby battery life.
#(8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
#(8020/8030 phones ignore this parameter if present no matter the SW version)
PROXY1_KEEPALIVE_SECS=14

# PROXYn_REREG_SECS specifies the the re-registration interval for a PROXY. This
# is the requested expiration interval sent in the REGISTER request message. It can
# be set to anything between 35 and 3600 seconds.
# The default value (which is applied if no parameter is supplied in this file) is
# 3600 which means the phone will attempt to register every 3580 seconds, if the
# server's response interval doesn't take precedence (see below).
#
# If the server has a lower maximum setting or a higher minimum setting that
# this value then the server's response will take precedence and this parameter
# will be ignored.
#
# The phone will always attempt to re-register with the PROXY 20 seconds prior to
# the expiration interval. Thus, if this parameter is set to 35 seconds, and the
# server's response interval doesn't take precedence, then the phone will attempt
# to re-register every 15 seconds.
#
# The lower this setting, the more the impact on the 8002's standby battery life.
#(8002 phones running version 130.003 and earlier code ignore this parameter if
# present)
#(8020/8030 phones ignore this parameter if present no matter the SW version)
PROXY1_REREG_SECS=60

# Favorites in the allusers file will be present in the favorites on all phones
# The username can be blank and can include escaped chars
# Useful features can be included such as call forwarding or dialing voicemail
#FAVORITE = "1234"; "Site Security"
#FAVORITE = "*98"; "Call Forwarding"

PKley,
All user file is used for when all or most users have something in common such as the PROXY1_ADDR, having this enabled in both the sip_xxx.cfg file would send multi PROXY1_ADDR to the TFTP server. The same with PROXY1_TYPE = TOSHIBA

1. Try Remarking the AllUser.cfg #PROXY1_ADDR
2. Also add a user password to the sip station in the CIX and in the sip_xxxx.CFG file
# AUTH = username; password
AUTH = 219; 219

Capture the TFTP log file on a failure.

Since it's only one phone, do I have to have the sip_allusers.cfg at all? I'm trying to eliminate as much as I can...

Yes, the set loads both after it verifies that there are no updated firmware.

Used some GREATLY streamlined .cfg files from a Polycom tech, phone connected but now I can't dial anything.
I'll enter digits and press Send but it just flips to a new screen with "enter more digits" and never does anything after that despite entering more digits...pressing send just clears what I entered.

pkley, the cfg files are specific to the CIX, if you did not get them from Toshiba its not going to work.
If you call Polycom back you can ask for their Toshiba 8002 Support person he will have the correct cfg files, otherwise General Support will not know this.