Page Comparison

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• Pricelist: associated pricelist for the correct calculation of calls cost (must be configured in WMS -> Trunks -> Pricelists)
• Title: description of the trunk
• Trunk name: trunk name
• Auth Login: provided by the VoIP Carrier for authentication
• From user: forced from number header and used for invite messages and for registration (if From domain is not empty), usually the same as Auth login
• From domain: forced from domain header and used in register and invite SIP messages
• Address or Host Name: address or host name of sip proxy; dynamic indicates that the trunk is used for incoming registration (incoming invites are allowed)
• Password: password for authentication provided by the VoIP Carrier
• Dialplan: Dialplan procedure for routing calls coming from this trunk (usually, “main”)
• Tone Zone: select the country/ region
• Country Code: used for number normalization, specify the code of the country where the trunk is used
• Keep-Alive: enable periodic sending of keep alive messages to the trunk

• Enable registration: enable outgoing registration (in case of PBXs SIP interconnection, it is enabled on the Client PBX SIP trunk and disabled on Server) 

• Registration proxy (appears after checking off Enable registration, optional): enter IP address or host name of a proxy server with the port (the default port is 5060) and credentials for access 
  
  

• Advanced

  • Audio codecs: enable the audio codecs supported by the trunk and the ptime values: 20ms – 40ms – 60ms (20ms by default); in case a different priority is needed, use  “Set” -> “Codec” Dialplan application
    

    Note
    Note: The default codec priority for SIP Trunks starting from WMS 6.02.20230306.1 is the following:  alaw ulaw g729

    
    

  • Video codecs: enable the video codecs supported by the trunk
  • T38: special parameters for T38 support and the maxdatagram
  • From number: add an expression to set a dynamic cid number for outgoing calls based on the office number or on the value “Set” -> “cid number” on trunk Dialplan application; works only if the operator allows setting of a cid number for outgoing calls
  • From name: add a regular expression to set a dynamic cid name for outgoing calls
  • Cid Header: add an additional header to set the cid number
  • Cid Body: indicate the contents of the cid header
  • Incoming CID: setting of a specific SIP header from which Cid header must be retrieved. Available options: p-asserted-identity,from (default) / from
  • Privacy Header: setting of the privacy header content supported by the operator to perform anonymous calls via feature code “Hide number” (92 by default)
  • Diversion / History-Info Header: add SIP header Diversion and History-info for outgoing INVITE to trunk in order to preserve the information on the original called number in case of call forwarding
    
    
  
  

  
  

  • Contact number: allows to set custom contact number in SIP trunk configuration; Note: The field is available starting from WMS v. 6.01.20221019.4
  • Show original caller number: can be used by operators which support displaying any number as cid, allows displaying original caller number from trunk in case of transfers/ mobility calls
  • Support Refer and Hold: allow the trunk to perform transfers and disable hold requests from the trunk

  • Play remote MoH: configure remote music on hold to be played instead of the local one for external calls
    

    Warning
    Limitation: For the Play remote MoH settings to apply, make sure the Support Refer and Hold option is enabled.

    
    

  • Session Timer: enable the check of the session validity to avoid pending calls; if enabled, the value 360secs is used, if disabled – 7200 secs
  • Force static SSRC: forbid SSRC change in RTP sessions; this option is recommended for some SIP operators which do not playback RTP packets after change of SSRC

  • Rport: the rport mechanism changes the SIP routing behavior, so that responses can be received through a NAT even if private addresses are used in the SIP headers; available options: INVITE (default), REGISTER/INVITE, off 

    Warning
    Limitation: If PBX is behind NAT and uses trunk with rport REGISTER, INVITE, the remote side may drop calls after 30 seconds.

    
    

  • Registration Expiry (sec): set the expiry time for outgoing register messages (default=600 sec / min=0)
  • Custom DNS Server: used to resolve the sip proxy domain name
  • Outbound proxy: all outgoing SIP messages are sent through the indicated host
  • NAT IP: enable the network connection IP
  • 100rel: enable reliable transmission of provisional messages
  • SDES-SRTP: enable SDES-SRTP protocol for calls through the trunk
  • Transport: select the transport protocol to be used: UDP/ TCP/ TLS/ auto (dns ptr-srv) 
  • DTMF mode: allows you to select the DTMF transfer mode: info/ rfc2833/ inband and specify the payload

  

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