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BRI/ PRI Media Gateways are EOL. |
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Introduction
Each trunk is displayed in WMS -> 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.
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.
Each trunk is displayed in the corresponding section of the page with the following information:
- For GSM trunk, the GSM signal power status is displayed
SIP Trunk Settings
Parameters:
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > sip-trunk-upd-1.jpg](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/sip-trunk-upd-1.jpg?version=1&modificationDate=1675440555615&cacheVersion=1&api=v2&width=400)
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > sip-trunk-upd-2.jpg](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/sip-trunk-upd-2.jpg?version=1&modificationDate=1675440594376&cacheVersion=1&api=v2&width=400)
- 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
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > sip-trunk-advanced-settings.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/sip-trunk-advanced-settings.png?version=1&modificationDate=1675440919726&cacheVersion=1&api=v2&width=400)
- 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
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
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
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > SIP-Trunk-Settings-2-upd.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/SIP-Trunk-Settings-2-upd.png?version=2&modificationDate=1675442193633&cacheVersion=1&api=v2&width=400)
Supported VoIP operators and examples of configuration
Check out Wildix supported VoIP providers in the USA, Italy, Germany, France, Switzerland, the Netherlands, Austria, Spain, the UK and examples of configuration on THIS PAGE.
Supported custom config parameters
You can set different custom config parameters for trunks, for example, set Q-Value for trunk registration or enable transcoding for web phone calls to trunks which do not support G711. Refer to Custom config parameters List for details.
BRI/ PRI Trunk Settings
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
- 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
- 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
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > ISDN-Trunk-Settings-1-upd.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/ISDN-Trunk-Settings-1-upd.png?version=1&modificationDate=1606206525000&cacheVersion=1&api=v2&width=400)
- Port X. Below you can set up parameters for a single port:
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > ISDN-Trunk-Settings-2-upd.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/ISDN-Trunk-Settings-2-upd.png?version=1&modificationDate=1606210234000&cacheVersion=1&api=v2&width=600)
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > ISDN-Trunk-Settings-3-upd.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/ISDN-Trunk-Settings-3-upd.png?version=1&modificationDate=1606210865000&cacheVersion=1&api=v2&width=600)
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 Settings
Parameters:
- Default Tone: select the country/ region
- 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
- 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
- Local Area Code: specify the area code
- Dialplan: 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)
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > GSM-Trunk-Settings-upd2.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/GSM-Trunk-Settings-upd2.png?version=1&modificationDate=1675765365905&cacheVersion=1&api=v2&width=550)
FXO Trunk Settings
Parameters:
- 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
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > FXO-Trunk-Settings-1-upd.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/FXO-Trunk-Settings-1-upd.png?version=2&modificationDate=1675442566825&cacheVersion=1&api=v2&width=650)
- 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
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > FXO-Trunk-Settings-2.2.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/FXO-Trunk-Settings-2.2.png?version=3&modificationDate=1675443035187&cacheVersion=1&api=v2&width=650)
- Disconnect signaling: select one of the options:
- Busy tone (the default value)
- Polarity reversal: enable/ disable this parameter to avoid dropped calls due to a wrong detection of polarity change on the copper line
- Open switching interval: enable/disable the parameter and set the Current Disconnect Threshold in ms
- Polarity reversal on answer: 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 second
These parameters are available for W04FXO R3 starting from WMS 5.03. |
![Documentation > ISDN - FXO - GSM - SIP Trunk Settings - Admin Guide > FXO-trunk-settings-disconnect-signaling-options.png](https://wildix.atlassian.net/wiki/download/thumbnails/30285495/FXO-trunk-settings-disconnect-signaling-options.png?version=2&modificationDate=1675443144877&cacheVersion=1&api=v2&width=650)
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