Show pagesource Old revisions Backlinks Export to PDF Rename Page ODT export Add to book Table of Contents Identities Howto's Defaults Commented File-structure Book Creator Add this page to your book Book Creator Remove this page from your book Manage book (0 page(s)) Help Identities Howto's Identity Audio Codecs Dialplan Defaults since 1.4 Commented File-structure identities.xml <!-- Identities type: list minimum: 6 elements maximum: 6 elements --> <identities> <!-- Describes a single identity --> <identity> <!-- enables the identity for registration type: boolean --> <active>false</active> <!-- The user part of a SIP URI to register the identity type: string --> <username>5551234</username> <!-- The password to authenticate the identity register type: string --> <password>secretPassword</password> <!-- The display name of the identity type: string --> <displayname></displayname> <!-- The SIP registrar hostname/address type: string --> <registrar>sip.example.org</registrar> <!-- The SIP registrar port type: integer minimum: 1 maximum: 65535 --> <registrarPort>5060</registrarPort> <!-- The domain of the identity (may differ from registrar) type: string --> <realm></realm> <!-- outgoing port of SIP messages. 0 will be used for generating a random one. type: integer maximum: 65535 --> <localSipPort>0</localSipPort> <!-- If true a SIP REGISTER will be send type: boolean --> <sendRegister>true</sendRegister> <!-- Timeout of registration type: integer minimum: 1 maximum: 60 --> <registerInterval>30</registerInterval> <!-- Settings that are needed for networks where a network address translation is active --> <nat> <!-- type: string accepted values: inactive activeStun active --> <useStunSip>inactive</useStunSip> <!-- type: string accepted values: inactive activeStun --> <useStunRtp>inactive</useStunRtp> <!-- Hostname/ip of the stun server type: string --> <stunUrl></stunUrl> <!-- port of the stun server type: integer minimum: 1 maximum: 65535 --> <stunPort>3478</stunPort> <!-- interval in which stun requests are issued. Value in minutes. type: integer minimum: 1 maximum: 60 --> <stunRequestInterval>5</stunRequestInterval> <!-- interval in which keepalive packets are send. Value in seconds. Used to ensure that every network component in use is reminded, that our connection is still active. type: integer minimum: 15 maximum: 255 --> <keepaliveInterval>45</keepaliveInterval> </nat> <!-- --> <outboundProxy> <!-- type: string accepted values: inactive automatic manual --> <mode>inactive</mode> <!-- type: string --> <url></url> <!-- type: integer minimum: 1 maximum: 65535 --> <port>0</port> <!-- type: string --> <url2></url2> <!-- type: integer minimum: 1 maximum: 65535 --> <port2>0</port2> </outboundProxy> <!-- rtp frame size in milliseconds (typically 10, 20 or 30 ms) type: integer minimum: 10 maximum: 30 --> <frameSize>20</frameSize> <!-- a set of rules, that are applied on outgoing calls and can modify the number, so that this number can be routed on the outside network. The order of the rules is important. The first matching rule is used. type: list minimum: 1 element --> <dialplan> <!-- if the pattern matches, the replace string is used. (...) defines blocks, that can be reused with $NUMBER. If three blocks are used in the pattern, we can use $1 $2 and $3 in the replace. --> <rule> <!-- type: string --> <pattern>555-(1234)</pattern> <!-- type: string --> <replace>$1</replace> </rule> </dialplan> <!-- Use the dialplan when starting a call from the CallLog. type: boolean --> <useDialplanInCallLog>false</useDialplanInCallLog> <!-- List of supported audio codecs. The order of the list is important. When creating a connection, the server and the client use the first codec, that is supported by both parties. type: list minimum: 1 element --> <audiocodecs> <!-- type: string accepted values: G.711 G.722 G.726 G.729 iLBC --> <audiocodec>G.722</audiocodec> </audiocodecs> <!-- "When clir is enabled we set the X-Privacy-Header in outgoing calls. We also change the From-Header to either:" "- not include a display-name" "- say the display-name is 'anonymous'" "- have the username set to ananymous, i.e. anonymous@registrar" type: string accepted values: displayEmpty displayAnonymous userAnonymous --> <clir>displayAnonymous</clir> <!-- The sip-username where the phone should subscribe to to get information about voice-messages. It is also used to make the call when trying to access these messages. type: string --> <vmb></vmb> <!-- plays local music on hold if a call of this identity is set on hold. When enabled, held calls will take away one audio-channel. Since our phones only have 2 channels, this will limit the number of simultaneous calls to two. type: boolean --> <localmoh>false</localmoh> <!-- Activates sips (SIP over TLS) on connections with this identity. Most likely you have to provide a certificate for the host you provided to be able to connect safely and successfully. type: boolean --> <secureConnection>false</secureConnection> <!-- type: string accepted values: mandatory optional disabled --> <srtp>optional</srtp> <!-- Validate the hostname against the CN of the provided Certificate. type: boolean --> <checkHostname>true</checkHostname> <!-- SIPS scheme is used if activated. Means the whole SIP path must be encrypted by TLS type: boolean --> <Peer2PeerTls>false</Peer2PeerTls> <!-- Root certificate used as a trust anchor for the host. Text in PEM format. type: string --> <certificate></certificate> <!-- used as pre code for the user part in an invite URI in case of a pickup szenario. e.g. '##06' type: string --> <pickupCode>##06</pickupCode> <!-- ip version (v4, v6 or automatic) for communication with the host type: string accepted values: IpV4 IpV6 IpAuto --> <ipVersion>IpAuto</ipVersion> <!-- The size of the incoming rtp buffer in milliseconds. Should be greater than twice the frameSize. Lesser values reduce delays in the audio handling. A greater value allows a more robust communication when there are network issues. type: integer minimum: 40 maximum: 160 --> <jitterBufferSize>60</jitterBufferSize> <!-- which of the phones interface should be used to connect to this identity. VPN requires a correctly setup and active vpn connection. type: string accepted values: network vpn --> <interfaceType>network</interfaceType> <!-- optional authentication username used by some of the providers type: string --> <authenticationUsername></authenticationUsername> <!-- protocol used for network traffic type: string accepted values: udp tcp --> <protocolType>udp</protocolType> <!-- use timeout for SIP sessions type: boolean --> <sessionTimerActive>true</sessionTimerActive> <!-- timeout for SIP sessions in minutes type: integer minimum: 2 maximum: 255 --> <sessionTimerValue>15</sessionTimerValue> <!-- the identity which this one is the fallback for (if the actual one cannot register) 0 := this is not a fallback identity type: integer maximum: 6 --> <fallbackFor>0</fallbackFor> </identity> </identities>