ISDN / FXO / GSM / SIP Trunk settings

You can find Start Guides of Wildix media gateways HERE

Each trunk is displayed in Trunks menu 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.

To add a SIP trunk, click the “+” 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.


Sip trunk

Parameters:

  • Pricelist: associated pricelist for the correct calculation of calls cost
  • Title: description of 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”)
  • Country Code: used for number normalization, specify the code of the country where the trunk is used
  • Enable registration: enables outgoing registration (in case of PBXs SIP interconnection, it is enabled on the Slave PBX SIP trunk and disabled on Master)
  • 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 the “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: allows to set 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 a cid number for outgoing calls
    • From name: allows to set a regular expression to set a dynaic cid name for outgoing calls
    • Cid Header: allows to set an additional header to set the cid number
    • Cid Body: indicates the contents of the cid header
    • Privacy Header: allows to set up the privacy header content supported by the operator to perform anonymous calls via feature code “Hide number” (92 by default)
    • 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: allows the trunk to perform transfers on the PBX
    • Session Timer: Enables the check of the session validity to avoid pending calls; if enabled, the value 360secs is used, if disabled – 7200 secs
    • 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
    • 100rel: enables reliable transmission of provisional messages
    • TCP Transport: enables TCP transport; if disabled, UDP is used
    • DTMF mode: allows you to select the DTMF transfer mode and specify the payload

Supported VoIP operators and examples of configuration

Check out Wildix supported VoIP providers in Italy, France and Germany and examples of configuration on THIS PAGE.


BRI/PRI trunk

Parameters:

  • Default Tone: select the country for the correct tone recognition
  • Country Code: used for number normalization (in case “Custom country” is selected, you can manually enter the country code)
  • Escape Pound (#) In SIP URI Username: enables support of pound (#) symbol in called number, normally used as a confirmation to send number; as a rule it is used in integrations with legacy PBXs
  • Clock Reference: port selection for clock management
  • Bearer Capability G711 / G729: used by the calling party to specify the kind of B channel that is being requested. it is indicated in the Q.931 SETUP message and it is used to distinguish among the different types of voice and data calls; in case “unrestricted” is selected, calls might be rejected if no end-to-end digital circuit is available
  • Port X: below you can set up parameters for a single port
    • Pricelist: the pricelist to calculate the costs for calls through the trunk
    • End point type: select between Terminal Equipment (connection to public ISDN) or 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 Master and Slave mode; in ISDN network one endpoint acts as a MASTER, generating the clock signal, and the other endpoints act as SLAVEs, synchronizing on the clock signal received from the MASTER; typically a NT type endpoint acts as clock MASTER, and a TE type endpoint acts as clock SLAVE; set to SLAVE for connection to operators
    • 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: 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
    • 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

Enbloc signaling for outgoing calls

Some routes are not enabled by the operator to support Enbloc method by default (when number is transmitted in a single message and not digit by digit, which means you first enter the number and then lift up the receiver).
In case this method doesn’t work, request your operator to enable enbloc signaling on your router.

Here is an example of Cisco 2811 router (Telecom Italia operator) configuration which enables the enbloc method:
dial-peer voice 102 pots
translation-profile incoming 373123456
answer-address .%373123456T
direct-inward-dial ***added line (riga aggiunta)***
port 0/2/0


gsm trunk

Parameters

  • Default Tone: select the country
  • Country Code: used for number normalization (in case “Custom countriy” is selected, you can manually enter the country code)
  • Gain RX VoIP: adjust the incoming gain level (VoIP channel)
  • Gain TX VoIP: adjust the outgoing gain level (VoIP channel)
  • Pricelist: selection of the pricelist for the line
  • Dialplan Procedure: Dialplan procedure for routing calls coming from this trunk (usually, “main”)
  • Number of sim: the number corresponding to the line which is present in the associated Dialplan procedure
  • SMS center: the option to insert manually the number of the SMS center
  • Enable SMS receiving: enable incoming SMS messages to the SIM card (it’s necessary to specify the e-mail address in the field below)
  • SMS2EMAIL service e-mail: e-mail address used for new messages notifications
  • Gain RX GSM: adjust the incoming gain level (GSM channel)
  • Gain TX GSM: adjust the outgoing gain level (GSM channel)
  •  



fxo trunk

Parameters

  • Default Tone: select the country
  • Country Code: used for number normalization (in case “Custom country” is selected, you can manually enter the country code)
  • Pricelist: the pricelist to calculate the costs for calls through the trunk
  • 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
  •  

Wildix W04FXO White Paper

 
 

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