en:products:comfortel-d-series:developer:provisioning:settings:identities

Identities

identities.xml
  1. <!-- Identities
  2. type: list
  3. minimum: 6 elements
  4. maximum: 6 elements -->
  5. <identities version="1.26.10">
  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. <!-- How to signal dtmf-digits:
  29. - rtpPayload: use event-packages in rtp-stream
  30. - inband: code tone-signal right into the rtp-stream
  31. - sipInfo: use sip-info messages
  32. type: enum
  33. accepted values:
  34. rtpPayload
  35. inband
  36. sipInfo -->
  37. <dtmfMethod>rtpPayload</dtmfMethod>
  38. <!-- The domain of the identity (may differ from registrar).
  39. There are scenarios, where the registrar is a specific host e.g sip.example.com
  40. but the sip participants are using a different domain e.g alice@example.org.
  41. In the above example, the domain would be example.org.
  42. type: string -->
  43. <domain></domain>
  44. <!-- If true a SIP REGISTER will be send
  45. type: boolean -->
  46. <sendRegister>true</sendRegister>
  47. <!-- Timeout of registration in minutes
  48. type: integer
  49. minimum: 5
  50. maximum: 60 -->
  51. <registerInterval>30</registerInterval>
  52. <!-- Timeout of subscriptions in minutes. Shorter values can produce a significant performance impact.
  53. type: integer
  54. minimum: 1
  55. maximum: 120 -->
  56. <subscriptionInterval>45</subscriptionInterval>
  57. <!-- What to do when a subscription fails. -->
  58. <subscriptionFailureHandling>
  59. <!-- When to retry: either never, or every xy seconds (linear) or quadratic (i.e. first wait xy seconds, if that retry fails wait 2 times xy seconds, if that fails wait 4 times as long until another attempt).
  60. type: enum
  61. accepted values:
  62. never
  63. linear
  64. quadratic -->
  65. <retryType>linear</retryType>
  66. <!-- Interval in seconds until a subscription is retried after it got terminated by remote. Different subscripton-failures use this setting differently. E.g. a notify with `Subscription-State: terminated` follows this setting 1:1 while a subscription-request that gets answered with `500 Server Internal Error` multiplies this setting by 10.
  67. type: integer
  68. minimum: 10 -->
  69. <retryInterval>180</retryInterval>
  70. </subscriptionFailureHandling>
  71. <!-- What to do when a registration fails. -->
  72. <registrationFailureHandling>
  73. <!-- When to retry: either never, or every xy seconds (linear) or quadratic (i.e. first wait xy seconds, if that retry fails wait 2 times xy seconds, if that fails wait 4 times as long until another attempt).
  74. type: enum
  75. accepted values:
  76. never
  77. linear
  78. quadratic -->
  79. <retryType>linear</retryType>
  80. <!-- Interval in seconds until a registration is retried after it failed.
  81. type: integer
  82. minimum: 1 -->
  83. <retryInterval>10</retryInterval>
  84. </registrationFailureHandling>
  85. <!-- Settings that are needed for networks where a network address translation is active -->
  86. <nat>
  87. <!-- type: enum
  88. accepted values:
  89. inactive
  90. active -->
  91. <useStunSip>inactive</useStunSip>
  92. <!-- type: enum
  93. accepted values:
  94. inactive
  95. active -->
  96. <useStunRtp>inactive</useStunRtp>
  97. <!-- 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.
  98. type: integer
  99. minimum: 15
  100. maximum: 255 -->
  101. <keepaliveInterval>45</keepaliveInterval>
  102. </nat>
  103. <outboundProxy>
  104. <!-- type: enum
  105. accepted values:
  106. inactive
  107. automatic
  108. manual -->
  109. <mode>inactive</mode>
  110. <!-- type: string -->
  111. <url></url>
  112. <!-- type: integer
  113. minimum: 1
  114. maximum: 65535 -->
  115. <port>5060</port>
  116. <!-- type: string -->
  117. <url2></url2>
  118. <!-- type: integer
  119. minimum: 1
  120. maximum: 65535 -->
  121. <port2>5060</port2>
  122. </outboundProxy>
  123. <!-- 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.
  124. type: list
  125. minimum: 1 element -->
  126. <dialplan>
  127. <!-- 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. -->
  128. <rule>
  129. <!-- a regular expression
  130. type: string -->
  131. <pattern>555-(1234)</pattern>
  132. <!-- type: string -->
  133. <replace>$1</replace>
  134. </rule>
  135. </dialplan>
  136. <!-- Use the dialplan when starting a call from the CallLog.
  137. type: boolean -->
  138. <useDialplanInCallLog>false</useDialplanInCallLog>
  139. <!-- for roaming users this setting marks wether or not a roaming user is logged in. Whenever a roaming user loggs out we clear the callists and other sensitive data. Though this setting is identity-based we do global clearing of all sensitive data.
  140. type: boolean -->
  141. <isLoggedInRoamer>false</isLoggedInRoamer>
  142. <!-- 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.
  143. type: list
  144. minimum: 1 element -->
  145. <audiocodecs>
  146. <!-- type: enum
  147. accepted values:
  148. G.711
  149. G.722
  150. G.726
  151. G.729
  152. iLBC
  153. speex
  154. opus -->
  155. <audiocodec>G.722</audiocodec>
  156. <audiocodec>G.711</audiocodec>
  157. <audiocodec>G.726</audiocodec>
  158. <audiocodec>G.729</audiocodec>
  159. <audiocodec>iLBC</audiocodec>
  160. <audiocodec>speex</audiocodec>
  161. <audiocodec>opus</audiocodec>
  162. </audiocodecs>
  163. <!-- When a sip-Message (an instant text-message) is received and shown: also sound an alert tone or not.
  164. type: boolean -->
  165. <messagesAlertWithSound>true</messagesAlertWithSound>
  166. <!-- "When clir is enabled we set the X-Privacy-Header in outgoing calls. We also change the From-Header to either:" "- say the display-name is 'anonymous'" "- have the username set to anonymous, i.e. anonymous@registrar"
  167. type: enum
  168. accepted values:
  169. displayAnonymous
  170. userAnonymous -->
  171. <clir>displayAnonymous</clir>
  172. <!-- Pbx sometimes uses P-Asserted-Identity headers to signal a transfer. This is usually not the case except
  173. for Auerswald PBXes. When set to true: phone re-interprets those headers as transfers if the target number
  174. is sufficiently different from the original number.
  175. type: boolean -->
  176. <pAssertedMightBeTransfer>true</pAssertedMightBeTransfer>
  177. <!-- 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.
  178. type: string -->
  179. <vmb></vmb>
  180. <!-- 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.
  181. type: boolean -->
  182. <localmoh>false</localmoh>
  183. <!-- 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.
  184. type: boolean -->
  185. <secureConnection>false</secureConnection>
  186. <!-- Use ICE to determine audio-connection. Disable this when ICE is not needed/used, this helps our SipStack to add the correct IP in sdp-offers, especially in VPN-scenarios.
  187. type: boolean -->
  188. <useIce>false</useIce>
  189. <!-- This option controls whether the IP address in SDP should be replaced with the IP address found in Via header of the REGISTER response, ONLY when STUN and ICE are not used. If the value is FALSE (the original behavior), then the local IP address will be used. If TRUE, and when STUN and ICE are disabled, then the IP address found in registration response will be used.
  190. type: boolean -->
  191. <sdpNatRewriteUse>true</sdpNatRewriteUse>
  192. <!-- This option controls whether the IP address in SDP should be determined by resolving the interface by the registar address.
  193. this can be useful if an registar is only available on a not primarily routed network, for example vpn
  194. type: boolean -->
  195. <sdpInterfaceDetection>false</sdpInterfaceDetection>
  196. <!-- type: enum
  197. accepted values:
  198. mandatory
  199. optional
  200. disabled -->
  201. <srtp>disabled</srtp>
  202. <!-- SIPS scheme is used if activated. Means the whole SIP path must be encrypted by TLS
  203. type: boolean -->
  204. <Peer2PeerTls>false</Peer2PeerTls>
  205. <!-- Root certificate used as a trust anchor for the host. Text in PEM format.
  206. type: string -->
  207. <certificate></certificate>
  208. <!-- used as pre code for the user part in an invite URI in case of a pickup szenario. e.g. '##06'
  209. type: string -->
  210. <pickupCode></pickupCode>
  211. <!-- When set, overwrites the country-specific dial-tone you hear when lifting the handset. Uses an internal tone-generator-syntax called CPT. e.g. 425@-6;(*/0/1).
  212. See http://wiki.auerswald.de/doku.php?id=en:products:comfortel-d-series:developer:callprogresstones
  213. type: string -->
  214. <dialTone></dialTone>
  215. <!-- ip version (v4, v6 or automatic) for communication with the host
  216. type: enum
  217. accepted values:
  218. IpV4
  219. IpV6
  220. IpAuto -->
  221. <ipVersion>IpAuto</ipVersion>
  222. <!-- optional authentication username used by some of the providers
  223. type: string -->
  224. <authenticationUsername></authenticationUsername>
  225. <!-- protocol used for network traffic
  226. type: enum
  227. accepted values:
  228. udp
  229. tcp -->
  230. <protocolType>tcp</protocolType>
  231. <!-- use timeout for SIP sessions
  232. type: boolean -->
  233. <sessionTimerActive>true</sessionTimerActive>
  234. <!-- timeout for SIP sessions in minutes
  235. type: integer
  236. minimum: 2
  237. maximum: 255 -->
  238. <sessionTimerValue>15</sessionTimerValue>
  239. <!-- the identity which this one is the fallback for (if the actual one cannot register) 0 := this is not a fallback identity
  240. type: integer
  241. maximum: 6 -->
  242. <fallbackFor>0</fallbackFor>
  243. <!-- ringtone assigned to this identity
  244. type: string -->
  245. <ringtone>SystemDefaultRingtone</ringtone>
  246. <!-- when a party wants two way early media, this setting sets the mic to muted.
  247. The mute will be released automatically when the call is connected.
  248. type: boolean -->
  249. <startEarlyMediaMuted>false</startEarlyMediaMuted>
  250. <!-- The sip-uri or number of an conference bridge. When configured conferences will be started in that bridge instead of locally.
  251. type: string -->
  252. <conferenceBridge>3344</conferenceBridge>
  253. <!-- when the dialscreen is opened, and a call will be created for this identity
  254. the input method will be set accoring to this setting
  255. type: boolean -->
  256. <startDialscreenAlphanumeric>false</startDialscreenAlphanumeric>
  257. <!-- This option is used to update the transport address and the Contact
  258. header of REGISTER request. When this option is enabled, the library
  259. will keep track of the public IP address from the response of REGISTER
  260. request. Once it detects that the address has changed, it will
  261. unregister current Contact, update the Contact with transport address
  262. learned from Via header, and register a new Contact to the registrar.
  263. This will also update the public name of UDP transport if STUN is
  264. configured.
  265. type: boolean -->
  266. <rewriteContact>true</rewriteContact>
  267. <!-- Specify the number of seconds to refresh the client registration
  268. before the registration expires.
  269. The value should be lower than the configured registration time.
  270. This setting is ignored when useHalvedExpiry is enabled.
  271. type: integer
  272. minimum: 5
  273. maximum: 3600 -->
  274. <earlyRefreshRegOffset>5</earlyRefreshRegOffset>
  275. <!-- Send re-register after the halved expiry duration elapsed
  276. type: boolean -->
  277. <useHalvedExpiry>false</useHalvedExpiry>
  278. </identity>
  279. </identities>
  • en/products/comfortel-d-series/developer/provisioning/settings/identities.txt
  • Last modified: 15.12.2022 11:29
  • by neubauers