Identities

identities.xml
  1. <!-- Identities
  2. type: list
  3. minimum: 6 elements
  4. maximum: 6 elements -->
  5. <identities>
  6. <!-- Describes a single identity -->
  7. <identity>
  8. <!-- enables the identity for registration
  9. type: boolean -->
  10. <active>false</active>
  11. <!-- The user part of a SIP URI to register the identity
  12. type: string -->
  13. <username>5551234</username>
  14. <!-- The password to authenticate the identity register
  15. type: string -->
  16. <password>secretPassword</password>
  17. <!-- The display name of the identity
  18. type: string -->
  19. <displayname></displayname>
  20. <!-- The SIP registrar hostname/address
  21. type: string -->
  22. <registrar>sip.example.org</registrar>
  23. <!-- The SIP registrar port
  24. type: integer
  25. minimum: 1
  26. maximum: 65535 -->
  27. <registrarPort>5060</registrarPort>
  28. <!-- The domain of the identity (may differ from registrar)
  29. type: string -->
  30. <realm></realm>
  31. <!-- outgoing port of SIP messages. 0 will be used for generating a random one.
  32. type: integer
  33. maximum: 65535 -->
  34. <localSipPort>0</localSipPort>
  35. <!-- If true a SIP REGISTER will be send
  36. type: boolean -->
  37. <sendRegister>true</sendRegister>
  38. <!-- Timeout of registration
  39. type: integer
  40. minimum: 1
  41. maximum: 60 -->
  42. <registerInterval>30</registerInterval>
  43. <!-- Settings that are needed for networks where a network address translation is active -->
  44. <nat>
  45. <!--
  46. type: string
  47. accepted values:
  48. inactive
  49. activeStun
  50. active -->
  51. <useStunSip>inactive</useStunSip>
  52. <!--
  53. type: string
  54. accepted values:
  55. inactive
  56. activeStun -->
  57. <useStunRtp>inactive</useStunRtp>
  58. <!-- Hostname/ip of the stun server
  59. type: string -->
  60. <stunUrl></stunUrl>
  61. <!-- port of the stun server
  62. type: integer
  63. minimum: 1
  64. maximum: 65535 -->
  65. <stunPort>3478</stunPort>
  66. <!-- interval in which stun requests are issued. Value in minutes.
  67. type: integer
  68. minimum: 1
  69. maximum: 60 -->
  70. <stunRequestInterval>5</stunRequestInterval>
  71. <!-- 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.
  72. type: integer
  73. minimum: 15
  74. maximum: 255 -->
  75. <keepaliveInterval>45</keepaliveInterval>
  76. </nat>
  77. <!-- -->
  78. <outboundProxy>
  79. <!--
  80. type: string
  81. accepted values:
  82. inactive
  83. automatic
  84. manual -->
  85. <mode>inactive</mode>
  86. <!--
  87. type: string -->
  88. <url></url>
  89. <!--
  90. type: integer
  91. minimum: 1
  92. maximum: 65535 -->
  93. <port>0</port>
  94. <!--
  95. type: string -->
  96. <url2></url2>
  97. <!--
  98. type: integer
  99. minimum: 1
  100. maximum: 65535 -->
  101. <port2>0</port2>
  102. </outboundProxy>
  103. <!-- rtp frame size in milliseconds (typically 10, 20 or 30 ms)
  104. type: integer
  105. minimum: 10
  106. maximum: 30 -->
  107. <frameSize>20</frameSize>
  108. <!-- 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.
  109. type: list
  110. minimum: 1 element -->
  111. <dialplan>
  112. <!-- 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. -->
  113. <rule>
  114. <!--
  115. type: string -->
  116. <pattern>555-(1234)</pattern>
  117. <!--
  118. type: string -->
  119. <replace>$1</replace>
  120. </rule>
  121. </dialplan>
  122. <!-- Use the dialplan when starting a call from the CallLog.
  123.  
  124. type: boolean -->
  125. <useDialplanInCallLog>false</useDialplanInCallLog>
  126. <!-- 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.
  127. type: list
  128. minimum: 1 element -->
  129. <audiocodecs>
  130. <!--
  131. type: string
  132. accepted values:
  133. G.711
  134. G.722
  135. G.726
  136. G.729
  137. iLBC -->
  138. <audiocodec>G.722</audiocodec>
  139. </audiocodecs>
  140. <!-- "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"
  141.  
  142. type: string
  143. accepted values:
  144. displayEmpty
  145. displayAnonymous
  146. userAnonymous -->
  147. <clir>displayAnonymous</clir>
  148. <!-- 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.
  149. type: string -->
  150. <vmb></vmb>
  151. <!-- 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.
  152. type: boolean -->
  153. <localmoh>false</localmoh>
  154. <!-- 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.
  155.  
  156. type: boolean -->
  157. <secureConnection>false</secureConnection>
  158. <!--
  159. type: string
  160. accepted values:
  161. mandatory
  162. optional
  163. disabled -->
  164. <srtp>optional</srtp>
  165. <!-- Validate the hostname against the CN of the provided Certificate.
  166.  
  167. type: boolean -->
  168. <checkHostname>true</checkHostname>
  169. <!-- SIPS scheme is used if activated. Means the whole SIP path must be encrypted by TLS
  170.  
  171. type: boolean -->
  172. <Peer2PeerTls>false</Peer2PeerTls>
  173. <!-- Root certificate used as a trust anchor for the host. Text in PEM format.
  174.  
  175. type: string -->
  176. <certificate></certificate>
  177. <!-- used as pre code for the user part in an invite URI in case of a pickup szenario. e.g. '##06'
  178. type: string -->
  179. <pickupCode>##06</pickupCode>
  180. <!-- ip version (v4, v6 or automatic) for communication with the host
  181. type: string
  182. accepted values:
  183. IpV4
  184. IpV6
  185. IpAuto -->
  186. <ipVersion>IpAuto</ipVersion>
  187. <!-- 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.
  188.  
  189. type: integer
  190. minimum: 40
  191. maximum: 160 -->
  192. <jitterBufferSize>60</jitterBufferSize>
  193. <!-- which of the phones interface should be used to connect to this identity. VPN requires a correctly setup and active vpn connection.
  194. type: string
  195. accepted values:
  196. network
  197. vpn -->
  198. <interfaceType>network</interfaceType>
  199. <!-- optional authentication username used by some of the providers
  200. type: string -->
  201. <authenticationUsername></authenticationUsername>
  202. <!-- protocol used for network traffic
  203. type: string
  204. accepted values:
  205. udp
  206. tcp -->
  207. <protocolType>udp</protocolType>
  208. <!-- use timeout for SIP sessions
  209. type: boolean -->
  210. <sessionTimerActive>true</sessionTimerActive>
  211. <!-- timeout for SIP sessions in minutes
  212. type: integer
  213. minimum: 2
  214. maximum: 255 -->
  215. <sessionTimerValue>15</sessionTimerValue>
  216. </identity>
  217. </identities>