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, implemented using different SIP accounts (SIP registrations) or routed via ISDN CAPI 2.0 line. The maximum supported number of Outgoing VOIP Call Routing rules is not limited.
The rules can be applied to one, several or all Virtual COM Ports (modems) simultaneously.
Fax Voip T.38 Modem 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. For example, if rule # 1 applies to all Virtual COM Ports (COM: *), and for any number dialed on (.*), then other rules # 2, # 3, etc. never be able to work under any conditions. Therefore, always pay attention to the preference of order of the Outgoing VOIP Call Routing rules.
If there is no one rule that satisfies the conditions of the call, the call can not be implemented. To make outgoing calls of selected type, 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
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 Outgoung VOIP Call Routing table and there is at least one SIP registration entry in the SIP registration table, Fax Voip T.38 Modem automatically invites to create default rules for Outgoing calls at the time of clicking <Apply> button.
The default rule is as follows:
To view the rules for separated Virtual COM Port (modem) line (for example COM20), click Ports and Modems=>Outgoing Plan (Outgoing Modem Call Routing) in the Fax Voip T.38 Modem Control Panel, then select the desired port from the list.
If you have problems with Outgoung VOIP Call Routing, you can activate Enable Call Routing Logging function. To do this click Options (Options and Logs) in the Fax Voip T.38 Modem Control Panel, then check Enable Call Routing Logging (saving and restarting are needed). Analysis of the log will help you to identify errors and to define correct rules for calls.
If the rules of some kind can not be used in the current configuration of the program, Fax Voip T.38 Modem 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.
Outgoung VOIP Call Routing Table contains the following information:
Rule
Unique number of the current rule.
From Line
This field describes the lines for which rule applies.
' COM:.* ' means "all Virtual COM Ports (modems)". 'COM:20,21' means that the rule applies only to lines associated with ports COM20 and COM21 and does not apply to another lines (COM22, COM23, etc.) Note that in the case of using of the native Fax Voip driver for Virtual Ports you see COM-ports connected to your Fax Program here. In the case of using of Third-party drivers for Virtual COM-port Pairs here you see COM-ports connected to the Fax Voip T.38 Modem side (In this case your Fax Program operates with another 'side' of Virtual COM-port Pair, thus your Fax Program works with other numbers of COM-ports).
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, H.323 or ISDN network. In the case of SIP, this field also indicates transport to be used (UDP, TCP or TLS) and whether Tel URI scheme or sips: prefix 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.
In the case of ISDN the following format is used: CAPI:[Controller]:[B-channel], where [Controller] is the # of ISDN Controller to be used ( ‘ * ‘ means first available controller) and [B-channel] is the # of B-channel to be used (‘ * ‘ means first available B-channel).
<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 T.38 Modem 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 SIP 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 SIP 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 SIP 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 field (usually FaxVoip). This method can be used if authantication is not required.
Case 5:
1. To SIP Line = 'sip.server.com'
2. Username = 'NAME'
3. There are no SIP Registrations 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 T.38 Modem 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.
In the case of ISDN, this parameter allows to override the default Caller ID Number. If you use *<default>, you should understand that default ISDN Username (number) (see in the ISDN CAPI 2.0 Settings) will be used in this case. Username consisting of letters is ignored, so digits should be used to include CallerID in the CAPI 2.0 message when doing outgoing call.
DisplayName
Name you would like to be reported to other users. DisplayName usually displayed at other side as the NAME part of Caller ID. This option overwrites the default Display Name (see in the VOIP Settings). *<> indicates, that the default 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, and it is used for outgoing SIP calls with the current rule.
Fax Mode
Displays Fax Mode applied to current rule. G711 means G711 fax (audio). T38 means support for T.38 mode, when sending a fax with this rule. T38+R indicates, that Fax Voip T.38 Modem sends T.38 re-invite without waiting the re-invite from the other party. To avoid possible problems with sending faxes, it is strongly recommended to pre-read the detail descriptions of possible Outgoing Fax Modes. (More details in the Chapter T38 and G711 (audio) Fax Modes of this manual).
G.711 MaxRate
Maximum bitrate for audio faxes (G711 fax mode). 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 Fax Settings. ‘ * ’ indicates, that the default Maximum bitrate value is used.
G.711 ECM
Possibility to use Error Correction Mode for audio faxes (G711 fax mode). This option overwrites the default ECM value, specified in the Fax Settings. ‘ * ’ indicates, that the default ECM value is used.
T.38 MaxRate
Maximum bitrate for T.38 faxes (T38 fax mode). T.38 MaxRate is always set to 14400.
T.38 ECM
Possibility to use Error Correction Mode for T.38 faxes (T38 fax mode). Error Correction Mode is always possible when T38 fax mode is used.
T.38 Redundancy
T.38 Redundancy in the format I/LS/HS, used with current rule. 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=>T.38 Redundancy” (in the case of SIP) or in the “H.323=>T.38=>T.38 Redundancy” (in the case of H.323). ‘ * ’ indicates, that the default T.38 Redundancy values are used.
Codecs
Codecs option overwrites the default codecs, specified in the “SIP=>Codecs=>Selected codecs” panel (in the case of SIP) or in the “H.323=>Codecs=>Selected codecs” panel (in the case of H.323). *<> indicates, that the default codecs are used. In the case of ISDN default G.711 A-law or G.711 u-law codec always used and can not be overridden via rule.
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.