Web Manager: Configuration Menus > System |
SIP Trunks |
This menu is used to add SIP trunks to the phone system configuration. The menu is accessed by selecting System in the menu bar and clicking SIP Trunks.
SIP Trunk Pre-Requisites
Before adding any SIP trunks, the system must be configured to support SIP operation:
• | SIP Trunk Channel Licenses The system can support 3 simultaneous SIP calls without needing licenses. Additional simultaneous calls, up to 20 in total, require the addition of licenses to the configuration. |
• | VCM Channels Note that for SIP calls the system also requires VCM channels. For a system those are provided by installing IP500 Combination base cards. Each of these cards (up to 2) provides 10 VCM channels. |
• | STUN Settings The system's STUN settings need to be configured to allow it to connect to the Internet for SIP calls. This is done through the STUN Settings for Network panel of the system's Advanced settings menu. |
SIP Trunk List
• | Descriptive Name A name for the trunk. This is affect the trunks operation. |
• | Domain Name: Default = Blank Each SIP Trunk configuration has a unique ITSP Domain name needed by SIP end points in order to register with the IP Office. This is a string which may be directly resolved to an IP Address, or may require DNS lookup to resolve the domain name to the Service provider’s address. If this field is left blank, registration is against the LAN IP address. |
• | Number of Channels: Default = 10 Number of trunk channels between 1 and 24. |
• | Authentication Name: Default = Blank. This value is provided by the SIP ITSP. |
• | Password: Default = Blank. This value is provided by the SIP ITSP. |
• | Details Clicking on Details will display the additional settings for the selected SIP trunk. |
This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on Details.
Trunk Parameters
• | Proxy Server Address In exceptional circumstances, the IP Address of the proxy server may be explicitly identified as either a different IP Address, or a different domain address resolvable by DNS. |
• | DNS Server Address If the proxy server address is set to a named server, the address of the DNS server used for name resolution should be entered here. |
• | Mobility Caller ID Format This option corresponds to the standard "draft-ietf-sip-privacy-04". The options are None, Remote Party ID, P Asserted ID or Diversion Header. |
• | Use Tel URI: Default = Off. Use Tel URI format (for example TEL: +1-425-555-4567) rather than SIP URI format (for example name@example.com). This affects the From field of outgoing calls. |
• | Check OOS: Default = On. Software level = 8.0+. When enabled, the system will regularly check if the trunk is in service. Checking that SIP trunks are in service ensures that outgoing calls are not delayed waiting for response on a SIP trunk that is not currently usable. Depending on the trunk's Transport Protocol, the trunks current service status is checked using the following methods: |
• | For all trunks, regular OPTIONS messages are sent. If no reply is received, the trunk is taken out of service. |
• | For TCP trunks, if the TCP connection is disconnected the trunk will be taken out of service. |
• | For trunks using DNS, if the IP address is not resolved or the DNS resolution has expired, the trunk is taken out of service. |
• | Call Routing Method: Default = Request URI. Software level = 8.0+. This field allows selection of which part of the incoming SIP information should be used for the incoming number. The options are to match either the Request URI or the To Header element provided with the incoming call. |
• | Association Method: Default = By Source IP address. Software level = 8.0+. This field sets the method by which a SIP line is associated with an incoming SIP request. The search for a line match for an incoming request is done against each line until a match occurs. If no match occurs, the request is ignored. This method allow multiple SIP lines with the same address settings. This may be necessary for scenarios where it may be required to support multiple SIP lines to the same ITSP. For example when the same ITSP supports different call plans on separate lines or where all outgoing SIP lines are routed from the system via an additional on-site system. |
• | By Source IP Address This option uses the source IP address and port of the incoming request for association. The match is against the configured remote end of the SIP line, using either an IP address/port or the resolution of a fully qualified domain name. This matches the method used by pre-8.0 systems. |
• | "From" header hostpart against ITSP domain This option uses the host part of the From header in the incoming SIP request for association. The match is against the line's Domain Name. |
• | R-URI hostpart against ITSP domain This option uses the host part of the Request-URI header in the incoming SIP request for association. The match is against the line's Domain Name. |
• | "To" header hostpart against ITSP domain This option uses the host part of the To header in the incoming SIP request for association. The match is against the line's Domain Name. |
• | "From" header hostpart against DNS-resolved ITSP domain This option uses the host part of the FROM header in the incoming SIP request for association. The match is found by comparing the FROM header against a list of IP addresses resulting from resolution of the line's Domain Name or, if set, the Proxy Server Address. |
• | "Via" header hostpart against DNS-resolved ITSP domain This option uses the host part of the VIA header in the incoming SIP request for association. The match is found by comparing the VIA header against a list of IP addresses resulting from resolution of the line's Domain Name or, if set, the line's Proxy Server Address. |
• | "From" header hostpart against ITSP proxy This option uses the host part of the “From” header in the incoming SIP request for association. The match is against the line's Proxy Server Address. |
• | "To" header hostpart against ITSP proxy This option uses the host part of the From header in the incoming SIP request for association. The match is against the line's Proxy Server Address. |
• | R-URI hostpart against ITSP proxy This option uses the host part of the Request-URI in the incoming SIP request for association. The match is against the line's Proxy Server Address. |
• | Calls Route Via Registrar: Default = On Normally SIP REGISTER requests and INVITE requests use the same server destination. This option should only be deselected when the service provider does not expect REGISTER requests to go to the same destination as the INVITE requests. You should only set this under specific instruction from the service provider. |
• | Separate Registrar This field is available when Calls Route Via Registrar is deselected. It is used to enter the address of the SIP server that should be used for registration. You should only set this under specific instruction from the service provider. |
• | Compression Mode: Default = Automatic Selection This defines the type of compression which is to be used for calls on this line. |
• | VOIP Silence Suppression: Default = Off When selected, this option will detect periods of silence on any call over the line and will not send any data during those silent periods. |
• | Call Initiation Timeout: Default = 4 seconds. Sets how long to wait for successful connection before treating the line as busy. |
• | RE-Invite Supported: Default = Off. When enabled, Re-Invite can be used during a session to change the characteristics of the session, for example when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk. Requires the ITSP to also support Re-Invite. |
• | DTMF Support: Default = RFC2833 This setting is used to select the method by which DTMF key presses are signaled to the remote end. The supported options are In Band, RFC2833 or Info. |
• | Use Offerer's Codec: Default = Off. Normally for SIP calls, the answerer's codec preference is used. This option can be used to override that behavior and use the codec preferences offered by the caller. |
• | Registration Expiry: Default = 60 minutes. This setting defines how often registration with the SIP ITSP is renewed following any previous registration. |
• | PRACK/100rel Supported: Default = Off. Software level = 8.0 This option sets whether Provisional Reliable Acknowledgement (PRACK) and 100rel are enabled. 100rel allows SDP negotiation to be completed while the call is in ringing state and allows further media changes for announcements or progress tones before a call is actually answered. PRACK, as defined in RFC 3262, provides a mechanism to ensure the delivery of provisional responses such as announcement messages. Provisional responses provide information on the status of the call request that is still in progress. |
• | Example: When a call to a mobile or cell phone is in the process of being connected, there may be a delay while the cell phone is located. Provisional information allow features such as an announcement "please wait while we attempt to reach the subscriber" to be played while the call setup is still in progress. |
• | Fax Transport Support: Default = Off. Software level = 8.0+ This option is only available if Re-Invite Supported is selected. When enabled, the system performs fax tone detection on calls routed via the line and, if fax tone is detected, renegotiates the call codec as configured below. The SIP line provider must support the selected fax method and Re-Invite. |
• | None Select this option if fax is not supported by the line provider. |
• | G711 G711 is used for the sending and receiving of faxes. |
• | T38 T38 is used for the sending and receiving of faxes. |
• | T38 Fallback T38 is used for the sending and receiving of faxes. On outgoing fax calls, if the called destination does not support T38, a re-invite it sent for fax transport using G711. |
• | Refer Support: Default = On. REFER is the method used by many SIP devices, including SIP trunks, to transfer calls. These settings can be used to control whether REFER is used as the method to transfer calls on this SIP trunk to another call on the same trunk. If supported, once the transfer has been completed, the IP Office system is no longer involved in the call. If not supported, the transfer may still be completed but the call will continue to be routed via the IP Office. |
• | Incoming: Default = Auto Select whether REFER can or should be used when an attempt to transfer an incoming call on the trunk results in an outgoing call on another channel on the same trunk. The options are: |
• | Always Always use REFER for call transfers that use this trunk for both legs of the transfer. If REFER is not supported, the call transfer attempt is stopped. |
• | Auto Request to use REFER if possible for call transfers that use this trunk for both legs of the transfer. If REFER is not supported, transfer the call via the system as for the Never setting below. |
• | Never Do not use REFER for call transfers that use this trunk for both legs of the transfer. The transfer can be completed but will use 2 channels on the trunk. |
• | Outgoing: Default = Auto Select whether REFER can or should be used when attempt to transfer an outgoing call on the trunk results in an incoming call on another channel on the same trunk. This uses system resources and may incur costs for the duration of the transferred call. The options available are the same as for the Incoming setting. |
This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on Details. Click on the edit icon in the Channel Setup panel.
Channel
Channel number, cannot be edited
• | Appearance ID Appearance ID numbers can be used to associate each channel a Line Appearance button on phones that support button programming. That button can then be used to make and answer calls using the channel. The line appearance ID for each channel is automatically assigned to those channels that have their Direction set as Bothways. |
• | Display Name: Default = Use Authentication Name This field sets the 'Name' value for SIP calls. |
• | Authentication Name When making a call, some service providers will often send an authentication challenge to validate the call before it is connected. This challenge requires the INVITE is re-submitted with Authentication data, including a network account name (provided by the service provider during installation). The network account name is the “Auth name”. It can be blank, in which case the Local URI is used. |
• | Password: Default = Blank. This value is provided by the SIP ITSP. |
• | Details Clicking on Details will display the additional settings for the selected SIP trunk channel. |
• | Appearance ID Appearance ID numbers can be used to associate each channel a Line Appearance button on phones that support button programming. That button can then be used to make and answer calls using the channel. The line appearance ID for each channel is automatically assigned to those channels that have their Direction set as Bothways. |
• | Direction: Default = Bothways Sets the allowed operation mode of the line. For systems running in Key mode, a line can be set to either Bothway (incoming and outgoing) or Incoming Call by Call (incoming only). For a system running in PBX mode, a line can be set to either Bothway (incoming and outgoing) or Call by Call (incoming and outgoing). |
• | Bothway When set to Bothway, incoming calls are presented to line appearance buttons matching the channels Appearance ID and to the channels Coverage Destination if set. For Key mode systems, outgoing calls are routed to the channel by pressing the matching line appearance button selection or by automatic line selection. In addition, on PBX mode systems, outgoing calls can be routed to the channel by including the line appearance in the ARS Selector that matches the dialed digits. |
• | Incoming Call by Call For systems running in Key mode, when set to Incoming Call by Call, incoming calls are routed using the Call by Call table. The Appearance ID, Coverage Destination and Ring Pattern fields are greyed-out as those settings are not applied. The trunk channel is not used for outgoing calls. |
• | Call by Call For systems running in PBX mode, when set to Call by Call, incoming calls are routed using the Call by Call table. The Appearance ID, Coverage Destination and Ring Pattern fields are greyed-out as those settings are not applied. In PBX mode, call by call entries can be used given ARS selector numbers (see below) which allow the trunk channel to also be used for outgoing calls. |
• | Local URI: The user part of the SIP URI. This specifies the contents of the FROM field when making a call (sending an INVITE). |
• | Anonymous: Withhold the calling parties information. |
• | Registration Required When selected, each local URI with unique Authentication credentials will register independently. |
• | P-Assert-ID If this field is configured, the channel can be used in SIPConnect Option 1 model for separating Public and Private PSTN identity (Sipconnect technical recommendation v 10, section 12.1.1). You can only use Explicit CLI configurations over SIP if using Option1 model for identity. In this case, calls over this channel will always have a fixed P-Assert-ID, but the From field may vary. |
• | Coverage Destination: Default = None. Phone Based Admin = Yes. This option sets where incoming calls should alert in addition to alerting on those extension that have a line appearance button programmed for the line. When the phone system is in night service mode, calls alert at the members of the Night Service group. |
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• | The Coverage Destination is not used for SIP trunks with their direction set to Incoming Call by Call. |
• | Ring Pattern: Default = 1. Phone Based Admin = Yes. Selects the ring pattern that should be used for calls when alerting on an extension. Calls forwarded, sent to call coverage or to a hunt group will always use the line ring pattern. Calls direct to an extension will use the line ringing pattern unless the user has Override Line Ringing set. Not used for calls presented to the user as a member of the Operator group. This feature is also not used for BST phones. |
• | VMS Delay - Day: Default = 2, Range = 0 to 6 (number of rings). Phone Based Admin = Yes. Set the number of rings before an unanswered call should be redirected the selected auto attendant when the system is not running in night service mode and the VMS Schedule is set to Always or Days Only. |
• | VMS Delay - Night: Default = 2, Range = 0 to 6 (number of rings). Phone Based Admin = Yes. Sets the number of rings before an unanswered call should be redirected to the selected auto attendant when the system is running in night service mode and the VMS Schedule is set to Always or Night Only. |
• | VMS Schedule: Default = Never. Phone Based Admin = Yes. This option determines when the VMS Delay settings above should be used and unanswered calls redirected to the selected auto attendant. The options are: |
• | Always Redirect calls when the system is in both day and night service modes. |
• | Day Only Redirect calls only when the system is not in night service. |
• | Night Only Redirect calls only when the system is in night service. |
• | Never Do not redirect calls. |
• | VMS Auto Attendant: Default = Auto Attendant 1. Phone Based Admin = Yes. This field allows selection of which auto attendant is used. |
This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on Details. Click on the edit icon in the Call by Call panel.
SIP Call By Call List
These settings are used to match calls received on SIP trunks channels set to Incoming Call-by-Call. For systems operating in Key System mode, the default entry is used for all calls for which there is no other match and is fixed to route those calls to the Operator Group.
• | ARS This setting is only shown for PBX mode systems. For those systems, each call-by-call entry can be assigned to an ARS Selector number. That selector number can then be used as the destination for outgoing calls. |
• | Local URI: The user part of the SIP URI. This specifies the contents of the FROM field when making a call (sending an INVITE). |
• | Destination Where incoming calls with matching digits should be routed. The drop-down list contains the extensions and groups on the IP Office system. |
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• | Authentication Name When making a call, some service providers will often send an authentication challenge to validate the call before it is connected. This challenge requires the INVITE is re-submitted with Authentication data, including a network account name (provided by the service provider during installation). The network account name is the “Auth name”. It can be blank, in which case the Local URI is used. |
• | Password: Default = Blank. This value is provided by the SIP ITSP. |
• | Details This control can be used to display additional settings associated with the call by call entry. |
• | Display Name: Default = Use Authentication Name This field sets the 'Name' value for SIP calls using this URI. |
• | P-Assert-ID If this field is configured, the channel can be used in SIPConnect Option 1 model for separating Public and Private PSTN identity (Sipconnect technical recommendation v 10, section 12.1.1). You can only use Explicit CLI configurations over SIP if using Option1 model for identity. In this case, calls over this channel will always have a fixed P-Assert-ID, but the From field may vary. |
• | Registration Required When selected, each local URI with unique Authentication credentials will register independently. |
This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on Details. Click on the edit icon in the Dial Plan panel.
The dial plan is used to apply number translations to the digits received by the line for output to the line provider and to indicate any special service required from the line provider, for example to withhold the call ID. The default dial plan is as shown below.
The following are the default entries used in a dial plan for North American locales.
Dialled Number |
Result |
Action |
xxxxxxxxxxN |
N |
Dial Local |
0N; |
0N |
Dial Local |
1N; |
1N |
Dial Local |
N; |
N |
Dial Local |
911 |
911 |
Dial Local |
*2xxN |
*2N |
Dial Local |
*3xxN |
*3N |
Dial Local |
*xxN |
*N |
Dial Local |
*65 |
|
Explicitly not Anonymous |
*67 |
|
Call Anonymously |
The default incoming number filter simply converts international USA numbers received into local 10 digit numbers. However, it is also useful for mapping PC calls (from skype, google, windows etc) into a dialable number plan. One nice way to use this is to map PC calls into numbers in area code "555".
This menu is accessed by selecting System and then SIP Trunks from the menu bar. Select the trunk required and click on the edit icon in the Dial Plan panel. Then click on the
edit icon in the Incoming Number Filter panel.
Incoming Number
Used to match the incoming number received.
• | Result The replacement for the incoming number. |
• | Include in Dial Plan When you select include in dial plan, the system will automatically substitute the number you dial for outgoing calls as well. |
© 2011 AVAYA - Issue 01.i.- 08:28, 24 November 2011 (sip_trunk_administration.htm) Performance figures, data and operation quoted in this document are typical and must be specifically confirmed in writing by Avaya before they become applicable to any particular order or contract. The company reserves the right to make alterations or amendments at its own discretion. The publication of information in this document does not imply freedom from patent or any other protective rights of Avaya or others. All trademarks identified by (R) or TM are registered trademarks or trademarks respectively of Avaya Inc. All other trademarks are the property of their respective owners. Last Modified: 21/09/2011 |