Snom Configuring snom phones for Mass Deployment Bedienungsanleitung

F REQUENTLY

A SKED

Q UESTION

Conguring 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 congure your

snom phone with cong 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

conguration 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 identication 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-specic 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 cong le followed

by a colon and its specic 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 cong 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 conguration

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>,

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 signicantly 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 unspecied 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 identied with the numbers 1

through 7. Line-specic 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 specied

<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>,

Default: Ringer1

Description: The ring tone to be used if a call comes to that specic 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_postx

Valid values: e.g. <proxy.company.de>

Default: blank

Description: Register postx, 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_sufx

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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