SIP Trunk Settings
This topic describes all the settings on a SIP trunk for reference.
Basic settings
- Basic
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Setting Description Name Give this trunk a name to help you identify it. Trunk Status Enable or disable the trunk. Select ITSP Template Select the country of your ITSP. Note: If no SIP trunk template is provided for your ITSP, select General.ITSP Select your ITSP from the list of certified SIP trunk providers.
- Detailed Configuration
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Setting Description Trunk Type Select a trunk type: - Register Trunk
- Peer Trunk
- Account Trunk
Register Trunk Transport Select the transport that is provided by the ITSP. Note: If you select TCP, make sure SIP TCP Port is enabled (Path: .Hostname/IP Enter the IP address or the domain of the ITSP. Port Enter the SIP port provided by the ITSP. Domain Enter the domain in SIP URI of a specific header like From, To header. Note: If the domain is not provided by ITSP, enter the same value as Hostname/IP.Username Enter the username to register to the ITSP. Password Enter the password that is associated with the username. Authentication Name Enter the authentication name to register to the ITSP. Enable Outbound Proxy If the trunk is configured to use an outbound proxy server, when users make outbound calls through this trunk, all the SIP packets will be sent to the outbound proxy server. Note: Contact your ITSP to check if they support outbound proxy, then configure outbound proxy settings under their guidance.Peer Trunk Transport Select the transport that is provided by the ITSP. Note: If you select TCP, make sure SIP TCP Port is enabled (Path: .Hostname/IP Enter the IP address or the domain of the ITSP. Port Enter the SIP port provided by the ITSP. Domain Enter the domain in SIP URI of a specific header like From, To header. Note: If the domain is not provided by ITSP, enter the same value as Hostname/IP.Account Trunk Transport Select the transport for a third-party device to register with. Note: If you select TCP, make sure SIP TCP Port is enabled (Path: .Username Specify a username for the trunk. Note: The username is regarded as the trunk number.Password Specify a password that is associated with the username. Authentication Name Specify an authentication name for a third-party device to register with.
Advanced settings
The advanced settings of VoIP trunk require professional knowledge of SIP protocol. Incorrect configurations may cause calling issues. It is wise to leave the default settings provided on the SIP trunk page. However, for a few fields, you need to change them to suit your situation.
The following settings are included on the Advanced page.
- Codec Setting
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Each newly created SIP trunk has a default preferred codec list. However, the default codec list may not match the codecs supported by your ITSP. To maximize the quality of calls and the amount of bandwidth used for calls, you can configure your preferred codec list to match the settings that your ITSP supports.
Yeastar P-Series Software Edition supports the following codecs:
- u-law
- a-law
- G729A
- GSM
- H264
- H261
- H263
- H263P
- iLBC
- G722
- G726
- SPEEX
- ADPCM
- MPEG4
- VP8
- Opus
- VoIP Setting
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Setting Description DTMF Mode Set the default mode for sending DTMF tones. - RFC4733 (RFC2833): DTMF will be carried in the RTP stream in different RTP packets rather than the audio signal.
- Info: DTMF will be carried in the SIP info messages.
- Inband: DTMF will be carried in the audio signal.
- Auto: The PBX will detect if the device supports RFC4733(RFC2833) DTMF. If RFC4733(RFC2833) is supported, PBX will choose RFC4733(RFC2833), or the PBX will choose Inband.
Qualify Enable this option to send SIP OPTION packet to SIP device to check if the device is up. Enable SRTP Enable or disable SRTP (encrypted RTP) for the trunk. T.38 Support Enable or disable T.38 fax for this trunk. Enabling T.38 will add the performance cost. We suggest that you disable T.38.
Inband Progress This Inband Progress setting applies to the extensions which make calls through this trunk. Note: To configure global Inband Progress setting, you need to contact Yeastar support to configure a custom config file.- Check this option: PBX will send a 183 Session Progress to the extension when told to indicate ringing and will immediately start sending ringing as audio.
- Uncheck this option: PBX will send a 180 Ringing to the extension when told to indicate ringing and will NOT send it as audio.
Ignore 183 Message without SDP This option determines the way PBX handles 183 messages without SDP. - Check this option: PBX will not forward 183 messages that don't contain SDP.
- Uncheck this option: PBX will process all the 183 messages without SDP to those with SDP and forward them.
Enable RTP Keep-alive Whether to send an RTP Comfort Noise (CN) frame. This helps to keep the NAT and firewall holes open during a call, so as to ensure the transmission of RTP traffic. Note:- This option is only available for register trunk and peer trunk.
- When this option is enabled, if the PBX does not send an RTP packet to the trunk within 1 second, an RTP Comfort Noise (CN) frame will be sent.
- Call Restriction
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Setting Description Call Restriction Type Specify based on which type of calls to restrict the max concurrent call number of this trunk. - Outbound Call: Only outbound calls will be restricted.
- All: Both outbound calls and inbound calls will be restricted.
Maximum Concurrent Calls Specify the maximum number of concurrent calls allowed in this trunk. The default is Unlimited.
- Trunk Network Interface Binding
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Specify which network interface the trunk will route through. You can bind the trunk's network route to the following options.
Note:- This feature is only available when Dual mode is selected as the Ethernet mode in basic network settings (Path: ), and is available for IPv4 SIP register trunk and SIP peer trunk only.
- If you make the configuration, you need to reboot the PBX system to take effect.
Setting Description Follow Static Routes This option indicates that the trunk's network will route follows the static routing rules of the PBX system. WAN This option indicates that the trunk's network will route through the WAN interface. Note: If you enable Number of WAN Ports in basic network settings (Path: ), you can select the specific WAN to associate with.
DIDs/DDIs
Direct Inward Dialling (DID), also called Direct Dial-in (DDI), is a service offered by telephone companies. For more information of DID concepts, see DID Number Overview.
- DID numbers are usually configured on inbound routes to distinguish inbound
calls.
For more information, see Route Inbound Calls based on DID Numbers.
- For more instructions on configuring the DID numbers, see Configure DID Numbers on a Trunk.
Inbound Caller ID Reformatting
When a user calls in the PBX, the trunk provider may send a caller ID that is inconvenient for you to redial directly.
In this case, you can reformat inbound caller ID based on a trunk. The caller ID will be reformatted before it is sent to the called party.
For more information, see Reformat Inbound Caller ID based on a Trunk.
Outbound Caller ID
Outbound caller ID is the phone number or name that is displayed on the called party's device.
You can set up a global outbound caller ID for a trunk or assign caller IDs for extension users.
If you set the caller ID number, when users make outbound calls through this trunk, the called party will see this caller ID number instead of the calling party's number.
For more information of outbound caller ID configurations, see Customize Outbound Caller IDs
SIP Headers
The SIP Headers settings require professional knowledge of SIP protocol. Incorrect configurations may cause calling issues. It is wise to leave the default settings provided on the SIP trunk page. However, for a few fields, you need to change them to suit your situation.
The following settings are included on the SIP Headers page.
- Inbound Parameters
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Parameter Description Get Caller ID From Decide from which header field will the trunk retrieve Caller ID. - Follow System
The trunk will follow the global Get Caller ID From setting.
- From
- Contact
- Remote-Party-ID
- P-Asserted Identify
- P-Preferred-Identity
Get DID From Different devices or providers may contain DID numbers in different SIP headers. When an inbound call through a SIP trunk reaches the PBX, the PBX needs to retrieve a correct DID number, or the call will fail. Adjust the setting after analysis of the SIP packets sent from the trunk provider. The following SIP headers are available to select:
- Follow System
The trunk will follow the global Get DID From setting.
- To
- Invite
- Diversion
- Remote-Party-IDNote: If this option is selected, but the SIP provider doesn't support Remote Party ID, the PBX will retrieve DID from INVITE header.
- P-Asserted Identify
- P-Called-Party-ID
- P-Preferred-Identity
- Follow System
- Outbound Parameters
- For outbound calls, you can define the parameters included in the
following SIP INVITE headers:
Parameter Description From User Part Define the caller ID that will be used of a SIP From header. For more information, see Options of outbound parameters.
From Display Name Part Define the caller ID name that will be used of a SIP From header. For more information, see Options of outbound parameters.
From Host Part Important:Define the domain or IP address to be used in the From field of a SIP INVITE header.- This parameter is only available for Peer Trunks.
- Set the parameter according to the requirements of your SIP trunk provider, otherwise, it may cause call issues.
- Default: Use the domain or IP address configured in the Domain field when creating the Peer Trunk.
- Custom: Use a custom domain or IP address. You can enter the custom value in the field next to the Custom drop-down list.
To Host Part Important:Define the domain or IP address to be used in the To field of a SIP INVITE header.- This parameter is only available for Peer Trunks.
- Set the parameter according to the requirements of your SIP trunk provider, otherwise, it may cause call issues.
- Default: Use the domain or IP address configured in the Domain field when creating the Peer Trunk.
- Custom: Use a custom domain or IP address. You can enter the custom value in the field next to the Custom drop-down list.
Diversion Optional: Define other parameters of a SIP INVITE header as needed. For more information, see Options of outbound parameters.
Remote-Party-ID P-Asserted-Identity P-Preferred-Identity P-Asserted-Identity URI Format Note: This parameter is available only when P-Asserted-Identity is not set to None.To assert and verify the identity of the caller, specify the format of P-Asserted-Identity:- SIP URI (sip:)
- TEL URI (tel:)
- Other Settings
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Setting Description User Agent If the ITSP requires User Agent for authentication, enter the User Agent information that is provided by the ITSP. Realm Realm is a string displayed to users so they know which username and password to use. Note: If you don't know what to fill in, contact your service provider for further instructions.Send Privacy ID Whether to send the Privacy ID in SIP header or not. The default is unchecked. User Phone Whether to add the parameter user=phone
as a request line in the header field of the SIP INVITE packet.Note: Enable this option only when the SIP provider requires.100rel Configure 100rel for this trunk. - Required: 100rel is required for this trunk.
- Supported: 100rel is supported by this trunk.
- Disabled: 100rel is disabled for this trunk.
Maxptime Select the value of the maxptime used when the PBX sends the INVITE packet. Note: If you select [Default], PBX will send a corresponding maxptime value according to the codec that is used for the outbound call.Support P-Early-Media Set whether the P-Early-Media field is included in the INVITE packet.