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The following Admin Guide describes Trunk Settings and explains how to set up various parameters. Updated: June 2018 WMS version: 3.88.41047.24 Permalink: https://confluence.wildix.com/x/8Qk8AQ |
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Introduction
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- 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: adds a regular expression to set a dynaic cid name for outgoing calls
- Cid Header: adds an additional header to set the cid number
- Cid Body: indicates 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: 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 from 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: 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): 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: 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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- Copy registration line for some trunk from /etc/callweaver/sip-registration.conf (Example: register => 144?144:123456:”144″@10.168.0.144:5060)
- Uncheck Enable registration option in SIP Trunk Settings (WMS -> Trunks):
- Add new line into /etc/callweaver/sip-general-custom.conf: register => 144?144:123456:”144″@10.168.0.144:5060~~0.6 (where 0.6 is q-value)
Run the command:
Code Block callweaver -rx “sip reload”
How to enable transcoding for web phone calls to trunks which do not support G711:
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To enable g729 transcoding, edit the file /etc/callweaver/sip-general-custom.conf:
[general]
- Add g729_transcoding=yes to [general] section.
BRI/ PRI Trunk Settings
Parameters:
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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)
- Cid Name via Facility IE (previously “Additional services”): enables the gateway to accept Facility messages and enables calling name delivery for US and Canada
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)
Note Note: Starting from WMS Version 3.88.41047.24, the option is no longer applied for trunks with the USA country code: 0/ 00 are not added to national/ international incoming 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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