Outgoing VOIP Call Routing
Depending on the number been dialed outgoing calls can be routed to different SIP or H.323 servers, to H.323 Gatekeeper or implemented using different SIP accounts (SIP registrations). The maximum supported number of Outgoing VOIP Call Routing rules is not limited.
Fax Voip FSP always checks Outgoing VOIP Call Routing rules, starting with the rule # 1. The call will be routed according to the first rule, which satisfies the conditions of the call. All other rules are ignored. If there is no one rule that satisfies the conditions of the call, the call can not be implemented. To make outgoing calls, you must have at least one rule in the Outgoing VOIP Call Routing table.
Depending on the number been dialed outgoing calls can be routed to
The rules of different types (SIP or H.323) highlighted in different colors in the Outgoing VOIP Call Routing table.
Outgoing SIP calls can be routed to
Outgoing H.323 calls can be routed to
The following translation rules can be applied to the dialed number:
The following call and fax settings can be set or overridden by the Outgoing VOIP Call Routing rule:
If there are no rules in the Outgoing VOIP Call Routing table and there is at least one SIP registration entry in the SIP registration table, Fax Voip FSP automatically invites to create default rule for Outgoing calls at the time of clicking <Apply> button.
The default rule is as follows:
If the rules of some kind can not be used in the current configuration of the program, Fax Voip FSP does not load them at startup and ignores these rules.
The rules that can not be used in the current configuration of the program are always highlighted in red in the Outgoing VOIP Call Routing table.
Outgoing VOIP Call Routing Table contains the following information:
Rule
Unique number of the current rule.
If Number =
Specifies the format of outgoing phone numbers for which this rule applies. For example ' .* ' means “all numbers”.
SIP/H.323
Specifies whether the outgoing call will be routed to the SIP or H.323 network. In the case of SIP, this field also indicates transport to be used (UDP or TCP) and whether Tel URI scheme is used in SIP INVITE message.
Dial on Number
Specifies Translation Rule for outgoing phone numbers. For example '<NUM>' means “without translation”, '9<NUM>' means “Prepend 9 before number”, '<NUM-2>' means “Remove first 2 digits”.
To Line
Displays SIP or H.323 server name or IP-address to which the call will be routed.
<GK> means that the call will be routed via H.323 Gatekeeper. In the case of H.323, if the registration with H.323 Gatekeeper is used, should understand that outgoing H.323 calls to arbitrary IP addresses may be prohibited by the Gatekeeper policy.
In the case of SIP, depending on the values of Username (see below) and some other Fax Voip FSP parameters the behavior can be different:
Case 1:
1. To Line = 'sip.server.com'
2. Username = '*<default>'
3. There is at least 1 entry in the SIP Registration table with
username@registrar = '<name>@sip.server.com', where <name> - any name.
The call will be routed using the first SIP Registration entry for the specified server.
Case 2:
1. To Line = 'sip.server.com'
2. Username = 'NAME'
3. There is an entry in the SIP Registration table with
username@registrar = '[email protected]'.
The call will be routed via SIP Registration entry '[email protected]'.
Case 3:
1. To Line = 'sip.server.com'
2. Username = 'NAME'
3. The server name is the same as the name of default proxy server (see in the Outbound Proxy Settings). The Username is the same as the Proxy Username. Use default Outbound Proxy option is checked.
The call will be routed to default Proxy. The username and password specified in the proxy settings can be used for authantication.
Case 4:
1. To Line = 'sip.server.com'
2. Username = '*<default>'
3. There are no SIP Registration entries with
username@registrar = '<name>@sip.server.com', where <name> - any name.
The call will be routed to sip.server.com with default SIP Username, specified in the SIP Settings (usually FaxVoip). This method can be used if authantication is not required.
Case 5:
1. To Line = 'sip.server.com'
2. Username = 'NAME'
3. There are no SIP Registration entries with
username@registrar = '[email protected]'.
The call will be routed to sip.server.com with username NAME. This method can be used if authantication is not required. Digital NAME usually displayed at other side as NUMBER part of Caller ID.
Username
In the case of SIP, Username (SIP-ID) that will be used with the rule. Usually coincide with the Username of one of the SIP registrations. If you use *<default>, you should understand that Fax Voip FSP can use the Username of one of the SIP registrations or default SIP Username, specified in the SIP Settings (usually FaxVoip) in this case. Different behaviors are considered in the description of To Line field. Digital Username usually displayed at other side as NUMBER part of Caller ID.
In the case of H.323, this parameter allows to override the default Caller ID Number. If you use *<default>, should understand that the top number from the Telephone numbers list (see in the H.323 settings) will be recognized by remote party as Caller ID Number.
DisplayName
Name you would like to be reported to other users. DisplayName usually displayed at other side as the NAME part of Caller ID. In the case of SIP, this option overwrites the default SIP Display Name, specified in the SIP Settings. In the case of H.323, this option overwrites the default H.323 Display Name, specified in the H.323 settings. *<> indicates, that the default Sip Display Name or H.323 Display Name is used.
Use Proxy
Displays the address of SIP Outbound Proxy, which is used with outgoing SIP calls (if specified). Using of different Outbound Proxy servers for SIP Registration and for Outgoing Calls via this registration is not recommended. Icon '=>' before the name of Outbound Proxy shows that you are using the Default Outbound Proxy setting (see in the Outbound Proxy Settings), and it is used for outgoing SIP calls with the current rule.
Fax Mode
Displays Fax Mode applied to current rule. G.711 means G.711 fax (audio). T.38 means support for T.38 mode, when sending a fax with this rule.
Audio Fax Tweaks
Individual Audio Fax options that override the general options that are specified in the Audio Fax Settings (SIP) or Audio Fax Settings (H.323).
Possible Audio Fax Tweaks options are:
MaxRate_out
Maximum bitrate for outgouing audio faxes (Fax over G.711 codec). Maximum bitrate value can be set 14400/9600/4800 which corresponds to rate limits of standard protocols used for facsimile. This option overwrites the default maximum bitrate, specified in the “SIP=>Audio Fax=>Audio Fax Transmission=>Maximum bitrate” field (in the case of SIP) or in the “H.323=>Audio Fax=>Audio Fax Transmission=>Maximum bitrate” field (in the case of H.323).
ECM_out (0 – disabled, 1 - enabled)
Possibility to use Error Correction Mode for outgouing audio faxes (Fax over G.711 codec). This option overwrites the default settings, specified in the “SIP=>Audio Fax=>Audio Fax Transmission=>Enable Error Correction Mode” (in the case of SIP) or in the “H.323=>Audio Fax=>Audio Fax Transmission=>Enable Error Correction Mode” (in the case of H.323).
T.38 Fax Tweaks
Individual T.38 Fax options that override the general options that are specified in the T.38 Fax Settings (SIP) or T.38 Fax Settings (H.323).
Possible T.38 Fax Tweaks options are:
T38MaxRate_out
Maximum bitrate for outgouing T.38 faxes. Maximum bitrate value can be set 14400/9600/4800 which corresponds to rate limits of standard protocols used for facsimile. This option overwrites the default maximum bitrate, specified in the “SIP=>T.38 Fax=>T.38 Fax Transmission=>Maximum bitrate” field (in the case of SIP) or in the “H.323=>T.38 Fax=>T.38 Fax Transmission=>Maximum bitrate” field (in the case of H.323).
T38ECM_out (0 – disabled, 1 - enabled)
Possibility to use Error Correction Mode for outgouing T.38 faxes. This option overwrites the default settings, specified in the “SIP=>T.38 Fax=>T.38 Fax Transmission=>Enable Error Correction Mode” (in the case of SIP) or in the “H.323=>T.38 Fax=>T.38 Fax Transmission=>Enable Error Correction Mode” (in the case of H.323).
T38SwitchOnTimer_out (Off – disabled, 0-20 – delay in seconds)
Forced switching to T.38 mode immediately (0 seconds) or after the specified time interval (1-20 seconds) for outgouing faxes. Fax Voip FSP sends T.38 re-invite to the other side without waiting for T.38 re-invite or CED tone from the other side. This option overwrites the default settings, specified in the “SIP=>T.38 Fax=>T.38 Fax Transmission=>Switch to T.38 after ... seconds” (in the case of SIP) or in the “H.323=>T.38 Fax=>T.38 Fax Transmission=>Switch to T.38 after ... seconds” (in the case of H.323).
T38SwitchOnCED_out (0 – disabled, 1 - enabled)
Forced switching to T.38 mode if the fax machine specific tone (CED signal) is detected when sending a fax. Fax Voip FSP sends T.38 re-invite to the other side without waiting for T.38 re-invite from the other side. This option overwrites the default settings, specified in the “SIP=>T.38 Fax=>T.38 Fax Transmission=>Switch to T.38 on CED tone received” (in the case of SIP) or in the “H.323=>T.38 Fax=>T.38 Fax Transmission=>Switch to T.38 on CED tone received” (in the case of H.323).
T38Redundancy
T.38 Redundancy in the format I/LS/HS. The values for (I)ndication, (L)ow (S)peed and (H)igh (S)peed IFP packets are specified separately. This option overwrites the default settings, specified in the “SIP=>T.38 Fax=>T.38 UDPTL Redundancy” (in the case of SIP) or in the “H.323=>T.38 Fax=>T.38 UDPTL Redundancy” (in the case of H.323).
Audio File
Audio file to be played before sending a fax when the outgouing fax call is answered. Overrides the default options that are specified in the Audio Settings.
The following commands are available in the Outgoing Plan Contextual Tab of the Ribbon:
Click to create new Outgoing VOIP Call Routing rule.
Click to create a copy of the selected Outgoing VOIP Call Routing rule. Can be useful when creating a large number of similar rules. To edit newly created rule, select it and use Edit command.
Click to edit Outgoing VOIP Call Routing rule. One of the entries should be selected. Alternatively you can double-click the selected entry.
Click to delete one or more Outgoing VOIP Call Routing rules. One or more entries should be selected.
Click to delete one or more Outgoing VOIP Call Routing rules. One or more entries should be selected.
Click to delete all rules in the list.
Move the selected rule up.
Move the selected rule down.
Click to select all the entries in the list.
Click to unselect all the entries in the list.
Click to invert the selected entries in the list.
Most of the commands placed on the Outgoing Plan Contextual Tab are also available from the context menu of the list.