Snom Configuring snom phones for Mass Deployment Instruction Manual
F REQUENTLY
A SKED
Q UESTION
Conguring snom phones
for Mass Deployment
Date: Feb-17-2004
Author: Sven Fischer
Document: faq-02-09-30-sf
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As stated in the FAQ „How can I update a snom phone?“, it is possible to congure your
snom phone with cong les, which consist of the following settings.
Phone settings
Settings are non-volatile name/value pairs that are stored in the ash of the phone. They
include registration information, dialed numbers, network settings and other information that
should be available after rebooting. Settings can be made read only on a per setting basis.
This is useful in environments where an operator sets up the phones and wants to avoid users
changing settings that affect the stability of the phone. In this way, expensive trouble ticket
searching can be reduced and the total cost of ownership of the phone be minimized.
Settings Files
Setting les are ASCII-based les containing lines (terminated with newline or carriage
return/newline pairs). Comments start with a # or a < character. The < and > characters allow
easy integration of html tags. Names may consist of the characters a-z, A-Z, 0-9 and _.
Appending a name with an “!” character means that this setting becomes writable by the
user. Appending the name with an “&” states that the setting is read only (this is the default).
If a setting is marked writable, the value in the settings le is only written to the phone if
that setting has not yet been set up at the phone. In order to overwrite such settings, do not
append the setting name with “!”.
A sample le looks like this:
<html>
<pre>
#This is a settings le
#
phone_name: testphone
dns_domain: intern.snom.de
dns_server1!: 192.168.0.9
</pre>
<html>
Downloading procedure
Settings are downloaded from web servers. HTTP is a very powerful way of getting
conguration information from anywhere in the world. The location of a le is described as a
URL, i.e. it begins with http://.
This is how the Settings are set up:
1. The settings that were stored in the ash memory are read.
2. Network and DNS is set up. This allows the phone to get settings from internet URLs.
3. It then checks the setting “setting_server” and loads the settings stored in the URL
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provided (e.g. http://www.snomag.de/snom200/snom200.htm). For a short description
of the le format, see below. The setting_server URL can also consist of a {mac}
statement like http://www.snomag.de/snom200/snom200.php?mac={mac}. The MAC
address is a unique identication ID of Ethernet devices. snom devices have the form
000413xxxxxx, where xxxxxx is a hexadecimal number identifying the snom device.
The phone automatically replaces the MAC address with its own mac address before
using the URL. Phone-specic settings can therefore be provided for each phone via
this mechanism. The “setting_server” setting can be set manually via webinterface or
automatically via DHCP (options 66 and 67) and through snom4S SIP proxy.
4. The de facto settings are added. These settings may differ from what has been set up
because DHCP may have changed them. The settings include:
• the IP address of the device,
• the net mask,
• the IP gateway,
• the hostname,
• the DNS domain, rst and second DNS server,
• the UTC offset in seconds,
• the time server,
• DHCP on or off
Each setting is described by its ‚Name:‘which should be added to the cong le followed
by a colon and its specic value. Name is followed by ‚Valid values:‘ where the valid values
are described. Each possible value is bracketed in <>.The brackets must be left out if you are
using the values stored in a cong le. ‚Default:‘ shows the value which is set by default after
you have removed all of the settings with a manual update or reset values! Lastly, ‚Description’
gives brief details of each setting. See also the snom4S SIP proxy manual for PNP conguration
via the proxy.
Redirection
redirect_event
Valid values: <all>, <busy>, <none>, <time>
Default: none
Description: Event that causes redirection. <all> redirects always, <none> never, <busy>
when the phone is in use and <time> after a timeout.
redirect_time
Valid values: e.g. <15>
Default: blank
Description: Number of seconds after which the phone will redirect the incoming call.
redirect_number
Valid values: e.g. <tb,sf> or <sip:[email protected];q=0.8,[email protected];q=0.9>.
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Default: blank
Description: The redirection target. If the proxy executes the redirection, this may be a
comma-separated list of alternate destinations (including probabilities).
redirect_busy_number
Valid values: e.g. <tb,sf> or <sip:[email protected];q=0.8,[email protected];q=0.9>.
Default: blank
Description: The redirection target when the redirect_event is set to busy.
Basic
language
Valid values: <English>, <English (UK)>, <Deutsch>, <Francais>, <Espanol>,
<Cestina>, <Nederlands>, <Polski>, <Portugues>, <Slovencina>, <Suomi>, <Svenska>,
<Japanese>, <Chinese>
Default: English
Description: Selects the language.
display_method
Valid values: <full_contact>, <display_name>, <user_name>
Default: full_contact
Description: Selects the display style of SIP URLs.
call_completion
Valid values: <true>, <false>
Default: false
Description: With <true> the phone offers after a “busy” the option to wait for the other
party to become free again and inform the user to make a new call attempt.
auto_dial
Valid values: <off>, <2> , <5>, <10>, <15>
Default: off
Description: Phone begins after 2, 5, 10 or 15 seconds to dial the number already typed
in.
block_url_dialing
Valid values: <true>, <false>
Default: false
Description: With <true> the user is no longer able to change the input type for editing
chars; it will remain numbers only.
challenge_response
Valid values: <true>, <false>
Default: true
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Description: <true> turns on displaying challenge request on the display to the user.
call_waiting
Valid values: <true>, <false>
Default: false
Description: If call waiting indication (CWI) is on (<true>), during a call parallel
incoming calls are signaled with a knocking sound and the caller hears a free signal. In
case of <false> only one incoming call in parallel can be handled and the caller hears a
busy signal.
cw_dialtone
Valid values: <true>, <false>
Default: false
Description: In the case of holding a party you will hear a free tone <true> or nothing
<false>
conf_hangup
Valid values: <true>, <false>
Default: false
Description: If you do not want the hook switch to cancel a call, it should be set to
<false>.
contrast
Valid values: Between <0> and <15>
Default: 8
Description: Determines the display contrast, but should not be used because each phone
reacts differently to depending for example on the temperature etc. It is better to set it
manually.
image_src
Valid values: <true>, <false>
Default: false
Description: The pictures for the internal web interface are taken from the phone (<false>)
or loaded via the Internet (<true>). Thus if you have a slow connection to the remote
phone, <true> will signicantly speed up the displaying of the phone‘s webpages.
guess_number
Valid values: <true>, <false>
Default: true
Description: Number guessing or auto completion while typing in numbers to dial.
deny_all_feature
Valid values: <true>, <false>
Default: true
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Description: If set to <true>, the “Deny All” option is offered on incoming calls. Otherwise
it is disabled.
admin_mode
Valid values: <true>, <false>
Default: true
Description: Administrator mode (all settings are accessible) or user mode (only a few
basic settings are accessible).
admin_mode_password
Valid values: Numbers and character strings of unspecied length, e.g. <1234>,
<0fg5gju>, <nhcndeve>
Default: 0000
Description: The password for entering the Administrator mode on the snom phone.
SIP Line Settings
The SIP stack on the phone supports seven lines, which are identied with the numbers 1
through 7. Line-specic settings are appended with the line number.
user_realname[1-7]
Valid values: e.g. <Jim Testa>
Default: blank
Description: This is the real name of the user that is displayed for line x.
user_name[1-7]
Valid values: e.g. for <sip:abc@company.de>, the user_name[1-7] would be abc.
Default: blank
Description: The account name for line x. This is a mandatory setting to set up an SIP
line!
user_host[1-7]
Valid values: e.g. for <sip:abc@company.de> the user_host would be <company.de>.
Default: blank
Description: The registrar for line x. This is also a mandatory setting to set up an SIP
line!
user_q[1-7]
Valid values: Values between <0.0> and <1.0>
Default: 1.0
Description: The probability of the registration for line x. This probability is used by some
proxies to call the registered phones one by one (sequential and parallel forking proxy).
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user_action[1-7]
Valid values: <none>, <proxy>, <redirect>
Default: none
Description: The action that should be performed by the registrar for line x. If specied
<redirect>, it will act as a redirect server. <proxy> will make it work as a proxy only.
user_expiry[1-7]
Valid values: <60>, <600>, <3600>, <7200>, <28800>, <86400>
Default: 86400 (one day)
Description: The proposed expiry time of the registration in seconds for line x.
user_mailbox[1-7]
Valid values: e.g. <abc> or <sip:[email protected]>.
Default: Blank
Description: The SIP url of the mailbox associated with line x.
user_ringer[1-7]
Valid values: <Ringer1>, <Ringer2>, <Ringer3>, <Ringer4>, <Ringer5>, <Ringer6>,
<Ringer7>
Default: Ringer1
Description: The ring tone to be used if a call comes to that specic line. See also “ring_
source”.
active_line
Valid values: [1-7]
Default: 1
Description: Number of the active SIP line. This is the line that is used as the originator of
an outgoing call.
SIP Stack Settings
sip_postx
Valid values: e.g. <proxy.company.de>
Default: blank
Description: Register postx, which is automatically chosen as registrar if the logon wizard
is used to logon an SIP user and the registrar has not been set up through the logon
wizard.
sip_proxy
Valid values: e.g. <proxy.company.de>, <192.168.0.24>
Default: blank
Description: Address or path of the outbound proxy that is used for calling.
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route_proxy
Valid values: <true>, <false>
Default: false
Description: Treat as initial route (<true>), <false> for address only
nat_detection
Valid values: <auto>, <upnp>, <stunclient>, <static>, <off>
Default: auto
Description: NAT detection according to your network.
stun_server
Valid values: e.g. <217.115.141.99:5062>
Default: blank
Description: IP address of a STUN server (hostname:port) if you have one.
stun_binding_interval
Valid values: Integer values e.g. 5, 10.
Default: blank
Description: STUN binding interval in seconds.
rtp_port_start
Valid values: Valid port numbers.
Default: blank
Description: First dynamic RTP port
rtp_port_end
Valid values: Valid port numbers.
Default: blank
Description: Last dynamic RTP port
network_id_name
Valid values: Valid name.
Default: blank
Description: Network identity (hostname)
network_id_port
Valid values: Valid port number.
Default: blank
Description: Network identity (port)
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use_nw_port
Valid values: <true>, <false>
Default: false
Description: Use local SIP port (<false>) or network port (<true>)
sip_retry_t1, sip_retry_t2
Valid values: Integer values, e.g. <500>.
Default: 500, 4000
Description: The time for re-sending SIP messages in milliseconds. They should be set to
500 and 4000 respectively.
session_timer
Valid values: Integer values, e.g. <2400>, <3200>.
Default: 3600
Description: Default time for session timer in seconds. 0 disables the session timer, 3600
is a reasonable value.
dirty_host_ttl
Valid values: Integer values, e.g. <60>, <120>.
Default: 0
Description: Time in seconds that the phone waits before repeating a non-answered
request.
max_forwards
Valid values: Integer values, e.g. <40>, <60>.
Default: 70
Description: The maximum number of hops allowed for an SIP request/response.
enum_sufx
Valid values: e.g. <e164.arpa>
Default: e164.arpa
Description: Route domain for ENUM, see FAQ “ENUM on snom phones”
tcp_threshold
Valid values: <auto>, <tcp>, <udp>
Default: auto
Description: Set transport layer for SIP messages either to TCP, UDP or automatic.
trace
Valid values: <true>, <false>
Default: true
Description: Enable tracing output of the current protocol stack.
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logon_wizard
Valid values: <true>, <false>
Default: false
Description: Logon wizard and mobility features available in snom key menu.
user_phone
Valid values: <true>, <false>
Default: true
Description: Use user=phone in SIP signaling.
use_mapped
Valid values: <true>, <false>
Default: true
Description: Use mapped address.
publish_presence
Valid values: <true>, <false>
Default: false
Description: Publish presence information on <true>.
refer_policy
Valid values: <auto>, <refer>, <bye>
Default: Auto
Description: Choose between REFER and BYE/Also method for initiating a transfer. Auto
makes an automatic selection depending on the capabilities indicated by the other party.
user_dp_str[1-7]
Valid values: dialplan regular expression, see snom4S proxy manual
Default: blank
Description: Up to 7 dialplan lines
sip_moh_server
Valid values: e.g. <sip:[email protected]>, <mh>, <192.168.0.40>
Default: blank
Description: SIP Address of music on hold server. The music is automatically played if the
phone is on hold.
SIP Authentication
auth_realm[1-9], auth_user[1-9], auth_pass[1-9]
Valid values: e.g. <company, 456876, answerIs42>
Default: blank
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