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Provisioning from a configuration file

It is possible to save a configuration xml file to be used as:

The structure of the xml file is as follows:

<locale>
	<region>
<recording>
	<enabled>
<logging>
	<level>
The phone application logs messages to log facility LOCAL0. Log level 1–5 (equivalent to Fatal–Trace)
	<log_sip>
Log SIP messages to log facility LOCAL1. Default is true.
	<remote_log>
Log messages to a remote log server. Default is false.
	<remote_host />
Remote log server.
<network>
	<net>
 
		<dhcp>
Specify if DHCP should be used to obtain network settings. If so, the other network settings won’t be used.
		<ip>
Specify the IP address of the Avaya B179.
		<netmask>
The netmask of the IP address.
		<gateway>
Specify the default gateway to be used.
		<dns>
		<dns>
Specify at most two Domain Name Servers to be used.
		<hostname>
Specify host name.
		<domain />
Specify domain name.
		<vlan>
 
			<enable>
Virtual LAN enabled if set to true.
			<id>
			<std_prio_map>
			<sip_priority>
			<media priority>
		<ether_8021x>
			<enable>
			<username />
			<eap_md5>
				<enable>
				<password>
			<eap_tls>
				<enable>
				<password>
	<qos>
		<dscp_sip>
		<dscp_media>
	<time>
		<ntp>
		<timezone>
		<daylight_save>
		<ntps>
VLAN ID.
<sip>
 
	<udp_transport>
Specify if UDP shall be used as transport.
	<udp_port>
Specify the UDP port to listen to.
	<tcp_transport>
Specify if TCP shall be used as transport.
	<tcp_port>
Specify the TCP port to listen to.
	<tls_transport>
Specify if TLS shall be used as transport.
	<sips_transport>
Specify if SIPS shall be used as transport.
	<tls_port>
Specify the TLS port to listen to.
	<rtp_port>
Specify the start port for RTP traffic.
	<outbound_proxy />
Specify the URL of outbound proxies to visit for all outgoing requests. The outbound proxies will be used for all accounts and will be used to build the route set for outgoing requests. The final route set for outgoing requests will consist of the outbound proxies and the proxy configured in the account.
	<use_stun>
Use Simple Traversal of UDP through NATs (STUN) for NAT traversal.

Default is No.

	<stun_domain />
Specify domain name to be resolved with DNS SRV resolution to get the address of the STUN servers. Alternatively, application may specify stun_host and stun_relay_host instead.
	<stun_host />
Specify STUN server to be used in ”HOST[:PORT]” format. If port is not specified, default port 3478 will be used.
	<use_turn>
Use Traversal Using Relay NAT (TURN) for NAT traversal. Default is no.
	<turn_host />
Specify TURN relay server to be used.
	<turn_tcp>
Use TCP connection to TURN server. Default is false.
	<turn_user />
TURN username.
	<turn_passwd />
TURN password.
	<nat_type_in_sdp>
Support for adding and parsing NAT type in the SDP to assist troubleshooting. The valid values are:

0: no information will be added in SDP and parsing is disabled

1: only the NAT type number is added

2: add both NAT type number and name

	<require_100rel>
Specify whether support for reliable provisional response (100rel and PRACK) should be required by default. Note that this setting can be further customized in account configuration.
	<use_srtp>
Specify default value of secure media transport usage. Note that this setting can be further customized in account configuration.

0: SRTP will be disabled, and the transport will reject RTP/SAVP offer.

1: SRTP will be advertised as optional and incoming SRTP offer will be accepted.

2: The transport will require that RTP/SAVP media shall be used.

	<srtp_secure_signaling>
Specify whether SRTP requires secure signalling. This option is only used when use_srtp option above is non-zero. Note that this setting can be further customized in account configuration.

0: SRTP does not require secure signalling

1: SRTP requires secure transport such as TLS

2: SRTP requires secure end-to-end transport (SIPS)

	<codec>
 
		<type>
		<name>
		<prio>
	<dtmf>
Codec type.

Codec name.

Codec priority (0–4)

DTMF signalling. Default is 2.

0: In-band

1: SIP message

2: RTP message

	<no_vad>
Disable VAD. Default is VAD enabled.
	<ec_tail>
Echo canceller tail length, in milliseconds.
	<enable_ice>
Enable ICE?
	<enable_relay>
Enable ICE relay?
	<enable_presence>
Enable the use of presence signalling.
<tls>
 
	<tls_password />
Password for the private key.
	<tls_method>
TLS protocol method from pjsip_ssl_method, which can be:

0: Default (SSLv23)

1: TLSv1

2: SSLv2

3: SSLv3

23: SSLv23

	<tls_verify_server>
Verify server certificate.
	<tls_verify_client>
Verify client certificate.
	<tls_require_client_cert>
Require client certificate.
	<tls_neg_timeout>
TLS negotiation timeout in seconds to be applied for both outgoing and incoming connections. If zero, no timeout is used.
<account>
 
	<valid>
If this account information is valid or not.
	<name>
User-defined name of the account.
	<id>
The full SIP URL for the account.
	<registrar>
This is the URL to be put in the request URI for the registration.
	<publish_enabled>
If this flag is set, the presence information of this account will be published to the server where the account belongs.
	<initial_auth>
If this flag is set, the authentication client framework will send an empty Authorization header in each initial request.
	<initial_algo />
Specify the algorithm to use when empty Authorization header is to be sent for each initial request (see above).
	<pidf_tuple_id />
Optional PIDF tuple ID for outgoing PUBLISH and NOTIFY. If this value is not specified, a random string will be used.
	<force_contact />
Optional URI to be put as Contact for this account. It is recommended that this field is left empty, so that the value will be calculated automatically based on the transport address.
	<require_100rel>
Specify whether support for reliable provisional response (100rel and PRACK) should be required for all sessions of this account.
	<proxy_uri />
Optional URI of the proxies to be visited for all outgoing requests that are using this account (REGISTER, INVITE, etc).
	<reg_timeout>
Optional interval for registration, in seconds. If the value is zero, default interval will be used.
	<cred>
Array of credentials. Normally, if registration is required, at least one credential should be specified to successfully authenticate the service provider. More credentials can be specified, for example when it is expected that requests will be challenged by the proxies in the route set.
		<realm>
Realm. Use ”*” to make a credential that can be used to authenticate any challenges.
		<scheme />
Scheme (e.g. ”digest”).
		<username>
Authentication name.
		<cred_data_type>
Type of data (0 for plaintext password).
		<cred_data>
The data, which can be a plaintext password or a hashed digest.
	<auto_update_nat>
This option is useful for keeping the UDP transport address up to date with the NAT public mapped address. When this option is enabled and STUN is configured, the library will keep track of the public IP address from the response of REGISTER request. Once it detects that the address has changed, it will unregister current Contact, update the UDP transport address and register a new Contact to the registrar.
	<ka_interval>
Set the interval for periodic keep-alive transmission for this account. If this value is zero, keep-alive will be disabled for this account. The keep-alive transmission will be sent to the registrar’s address after successful registration.
	<ka_data />
Specify the data to be transmitted as keep-alive packets. Default: CR-LF.
	<use_srtp>
Specify whether secure media transport should be used for this account.

0: SRTP will be disabled and the transport will reject RTP/SAVP offer.

1: SRTP will be advertised as optional and incoming SRTP offer will be accepted.

2: The transport will require that RTP/SAVP media is used.

	<srtp_secure_signaling>
Specify whether SRTP requires secure signalling. This option is only used when use_srtp option above is non-zero.

0: SRTP does not require secure signalling

1: SRTP requires secure transport such as TLS

2: SRTP requires secure end-to-end transport (SIPS)

<account>
Same as above, but for account 2.
<provisioning>
	<upgrade>
 
		<url>
Place to find software upgrades. The supported URL types are: HTTP, FTP, and TFTP.
	<dev_mgnt>
 
		<enable>
Device management enabled, true or false.
		<use_dhcp_option>
Use DHCP option for DM server address.
		<dhcp_option>
Specification of which DHCP option to use.
		<file_server_address>
DM server address if not provided by DHCP option.
		<pagename />
Base name of configuration files to download.
		<type />
Configuration file type specification.
		<update_interval>
Timing for downloading files. Shall be entered in crontab format: * * * * * where the * stands for minute (0–59), hour (0–23), day of month (1–30), month (1–12), day of week (0–7) (Sunday =0 or 7)

Example: 0 6 * * * = the files are downloaded daily at 6:00.

		<https_check_srv_cert>
Controls server certificate, true or false.
		<https_protocol>
Possibility to set https protocol if open-ssl auto detection fails.
<www>
 
	<enable_https>
Secure communication to the Avaya B179 web server. Default is false.
<pa>
 
	<enable_pa>
PA enabled, true or false.
	<enable_internal_mic>
Internal mic enabled when PA set to true.
	<enable_internal_spkr>
Internal speakers enabled when PA set to true.
	<calibration>
Calibration value. Note that 0 is auto, 1 is calibration value 1, 2 is calibration value 1, etc.
<ldap>
 
	<enable>
LDAP enabled, true or false.
	<name_filter>
Name filter according to RFC2254.
	<server_url>
LDAP server address.
	<search_base>
The DN (distinguished name) of the search base.
	<username />
	<password />
	<max_hits>
	<country_code>
	<area_code>
	<external_prefix />
	<min_length_for_ext_prefix />
	<exact_length_for_no_ext_prefix />
	<number_prefix_for_no_ext_prefix />
	<number_attributes>
	<display_name>
	<sort_results>