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Introduction

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• 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
  • 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
    
    

  
  

  • 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
  • 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: forbids 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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• Tone Zone: select the country/ region
• Country Code: used for number normalization. In case “Custom country” is selected, you can manually enter the country code
• Impedance: select one of the impedance values to avoid noise during line connection
• DC Impedance (disabled by default)
• Obtain FSK CID from (available for W04FXO 2020): select the value to display the correct Caller ID. The default value depends on a language selected in WMS Settings -> PBX -> Language&Region:
  • for English-US, the default value is Callee
  • for other languages, the default value is Nu

• Pricelist: the pricelist to calculate the costs for calls through the trunk
• Local Area Code: specify the area code
  
• Dialplan: Dialplan procedure for routing calls coming from this trunk (usually, “main”)
• Hotline Number: the number corresponding to the line which is present in the associated Dialplan procedure
• Detect caller ID (available for FXO gateways pre 2020): enable/disable caller ID detection
• Caller ID type (available for FXO gateways pre 2020): select the standard for transmitting the caller ID information
• Flash time (available for FXO gateways pre 2020): timeout (in msec) for hookflash detection
• Dialing answer delay time: interdigit dial timeout (in sec), when terminated, the call is sent to trunk (the value cannot be less than “1”)
• FAX type: select the mode (T38 or Bypass)
• Gain RX FXO: adjust incoming audio gain
• GAIN TX FXO: adjust outgoing audio gain
• Hang up cause:
  • Polarity reversal detection: enable/ disable this parameter as the cause for disconnecting the call
  • Silence detection: enable/ disable this parameter as the cause for disconnecting the call and specify the timeout in seconds
    
    

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