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Introduction

Each trunk is displayed in WMS -

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> Trunks in the corresponding section (SIP, BRI/PRI, GSM/UMTS, FXO) with real time registration status. For GSM gateways the signal power status is also displayed.

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SIP trunks must be added manually. To add a SIP trunk, click “+” icon below the SIP trunk table.

GSM, BRI/PRI and FXO trunks appear in the corresponding tables of the Trunks menu after you have provisioned the gateways in the Devices menu of the WMS.

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Installation Guides: W02/04BRI, W01/02PRI, W01GSM, W04FXO & W04FXO White Paper.

Updated: March 2018

Permalink: https://confluence.wildix.com/x/8Qk8AQ

SIP Trunk Settings

Parameters:

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Each trunk is displayed in the corresponding section of the page with the following information:

• Country code

• SIP registration status (outgoing or incoming, depending on the SIP trunk configuration):

  • Green: trunk is registered

  • Grey: trunk is not registered

• For GSM trunk, the GSM signal power status is displayed

• For BRI/BRI and GSM trunks, status of Layer 1 and Layer 2 for each port is displayed:

• Green: active

• Red: error (inactive)

• Grey: no event received (inactive)

SIP Trunk Settings

Parameters:

• 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: enables periodic sending of keep alive messages to the trunk
• Enable registration: enables 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: enables 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: enables the video codecs supported by the trunk
  • T38: special parameters for t38 support and the maxdatagram
  • From number: adds 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: allows to set adds a regular expression to set a dynaic cid name for outgoing calls
  • Cid Header: allows to set adds an additional header to set the cid number
  • Cid Body: indicates the contents of the cid header
  • Incoming CID:  allows to set setting of a specific sip SIP header from which Cid header must be retrieved. Available options: from / p-asserted-identity,from (default) / from
  • Privacy Header:  allows to set up setting of the privacy header content supported by the operator to perform anonymous calls via feature code “Hide number” (92 by default)
  • Diversion Header Diversion / History-Info Header:  allow to indicate the intended recipient of a adds 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: allows the trunk to perform transfers and disables hold requests on the PBXfrom trunk
  • Session Timer: enables the check of the session validity to avoid pending calls; if enabled, the value 360secs is used, if disabled – 7200 secs
  • Force static SSRC: enables the source to identify the source of a streamRport: select INVITE, REGISTER/INVITE, off options in case PBX is located behind NATforbids 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): sets 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: enables the network connection IP
  • 100rel: enables reliable transmission of provisional messages
  • Transport: enables one of the valuesselect 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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• Port X. Below you can set up parameters for a single port:
  • Pricelist: the pricelist to calculate the costs for calls through the trunk
  • Local area code: specify the area code
  • End point type: select between TE (Terminal Equipment) (connection to public ISDN) or NT (Network Termination) mode (connection to ISDN phones or other devices in TE mode, e.g. PBXs)
  • Dialplan: Dialplan procedure for routing calls coming from this trunk (usually, “main”)
  • Clock: select between Generate clock and Use provided clock mode; in ISDN network one endpoint acts as a SERVERr, generating the clock signal, and the other endpoints act as CLIENTs, synchronizing on the clock signal received from the SERVER; typically a NT type endpoint acts as clock SERVER, and a TE type endpoint acts as clock CLIENT; set to CLIENT for connection to operators
  • Line Coding: select one of the line coding schemes: B8ZS (Bipolar 8-Zero Substitution), HDB3 (High Density Bipolar 3), AMI (Alternate Mark Inversion)
  • Network type: select the type of the ISDN network
  • Connection type: select the type of ISDN connection between point-to-point and point-to-multipoint
  • Link Establishment: select the connection establishment type between permanent and on demand
  • Signaling Protocol: select the signaling protocol between asymmetric DSS1 (used for connection to public ISDN) and symmetric QSIG (used for trunking between several PBXs)
  • Additional services: enables the gateway to accept Facility messages

  • Overlap dialing (enabled by default): enables the gateway to begin processing a call as soon as it can determine a destination from dialed digits that form only part of a complete number; the feature must be enabled for DID (Direct Inward Dialing) management

    Warning
    Important: The option may need to be disabled for some VoIP Carriers for make outgoing calls due to the differences in D channel signaling performed for this option.

    
    

  • Inband DTMF Dialing: enables recognition of inband DTMF tones
  • Send Restart On Startup: enables sending of restart request to the operator each time the gateway reboots
  • Calling Name Max Length: specify the max length of the calling name
    
    

  
  

  • Channel Allocation Strategy: select the channel allocation strategy for this port
  • Sending Complete: includes Sending Complete information element into setup message; in this case the remote endpoint does not wait for digits coming in overlap mode
  • Progress Indicator: enables the gateway to provide the tone of progress during the indicated operations; normally these parameters are disabled
  • Maximum Facility Waiting Delay (ms): maximum waiting time of a Facility request
  • Use Implicit Inband Info: enables use of tones coming from endpoints
  • Correct incoming calling numbers: used for correct caller number visualization (adds 0 to national and 00 to international calls)
  • Signal Information Element: enables the use of Signal Information Element field to provide tones during different stages of call processing; this parameter must be enabled in NT mode only
  • Enable CNG / CED Tone Detection: allows voice and fax calls on the same line; enables the gateway to detect calling tone (CNG) generated by a fax machine, and called (answering) tone (CED) to enable T38 protocol

  • Detection Threshold: set up the threshold level for FAX detection

  
  

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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): 
  
  

• 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: enable/disable caller ID detection
• Caller ID type: select the standard for transmitting the caller ID information
• Flash time: 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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