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Navigation: System Settings > Line > Add/Edit Trunk Line > SIP Line > SIP Line
These settings are mergeable with the exception of the Line Number setting. Changing the Line Number setting requires a “merge with service disruption”. When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line are dropped.
Offline editing is not required.
Field |
Description |
|---|---|
Line Number |
Default = Auto-filled. Range = 1 to 249 (IP500 V2)/1 to 349 (Server Edition). The line number must be unique for each line in the configuration. IP500 V2 systems reserved line numbers 1 to 16 for internal hardware. |
ITSP Domain Name |
Default = Blank. This field is used to specify the default host part of the SIP URI in the For the user making the call, the user part of the From SIP URI is determined by the settings of the SIP URI channel record being used to route the call (see SIP Line > SIP URI > Local URI). This will use one of the following:
For the destination of the call, the user part of the To and R-URI fields are determined by dial short codes of the form 9N/N"@example.com” where N is the user part of the SIP URI and "@example.com" is optional and can be used to override the host part of the To and R-URI. |
Local Domain Name |
Default = Blank. An IP address or SIP domain name as required by the service provider. When configured, the Local Domain Name value is used in
If both the ITSP Domain Name and Local Domain Name are configured, Local Domain takes precedence. Local Domain Name is not used in the |
URI Type |
Default = SIP URI. Set the format the IP Office uses for SIP URI entries in headers.
|
Location |
Default = Cloud. You can set Location values for the IP Office system and for individual extensions and lines. Associating a line with a location:
|
Prefix National Prefix International Prefix Country Code |
The IP Office uses these values to adjust incoming numbers to match the format required for outgoing calls and used in system directory entries.
|
Name Priority |
Default = System Default. For SIP trunks, the caller name displayed on an extension can either be that supplied by the trunk or one obtained by checking for a number match in the extension user's personal directory and the system directory. This setting determines which method is used by the line. The options are:
|
Description |
Default = Blank. Maximum 31 characters. You can use this field to enter a description for the configuration entry. The description is not used elsewhere. |
Network Type |
Default = Public. This option is available when System | Telephony | Telephony | Restrict Network Interconnect is enabled. It allows you to configure trunks as either Public or Private.
|
In Service |
Default = On. When this field is not selected, the SIP trunk is unregistered and not available to incoming and outgoing calls. |
Check OOS |
Default = On. If enabled, the system will regularly check if the trunk is in service using the methods listed below. Checking that SIP trunks are in service ensures that outgoing call routing is not delayed waiting for response on a SIP trunk that is not currently usable. For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an OPTIONS message is received the trunk is 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. |
Field |
Description |
|---|---|
Refresh Method |
Default = Auto. The options are: Auto, Reinvite or Update. When Auto is selected, if UPDATE is in the Allow: header from the far SIP endpoint, then it is used. Otherwise INVITE is used. |
Timer (seconds) |
Default = On Demand. Range = 90 to 64800 This field specifies the session expiry time. At the half way point of the expiry time, a session refresh message is sent. When set to On Demand, IP Office will not send a session refresh message but will respond to them. |
Redirection and blind transfer are configured separately. By default, they are disabled.
A supervised transfer occurs when a consultation call is made and the REFER contains a Replaces: header indicating the CallID of another call leg which the REFERing agent has already initiated with the REFER target.
Do not change these settings unless directed to by the SIP service provider.
Field |
Description |
|---|---|
Incoming Supervised REFER |
Default = Auto. Determines if IP Office will accept a REFER being sent by the far end. The options are:
|
Outgoing Supervised REFER |
Default = Auto. Determines if IP Office will attempt to use the REFER mechanism to transfer a party to a call leg which IP Office has already initiated so that it can include the CallID in a Replaces: header. The options are:
|
Send 302 Moved Temporarily |
Default = Off. A SIP response code used for redirecting an unanswered incoming call. It is a response to the INVITE, and cannot be used after the 200 OK has been sent as a response to the INVITE. |
Outgoing Blind REFER |
Default = Off. When enabled, a user, voicemail system or IVR can transfer a call by sending a REFER to an endpoint that has not set up a second call. In this case, there is no Replaces: header because there is no CallID to replace the current one. This directs the far end to perform the transfer by initiating the new call and release the current call with IP Office. |