Cactus

VoIP PBX Software - Shamrock Software GmbH

SIP server

This configuration dialogue controls the communication between IP telephones and the Cactus server with the Session Initiation Protocol SIP.

• Realm: Internal realm of your Cactus server, e.g. "Cactus". Not all IP phones have a realm configuration field, so Cactus allows the internal or external IP address for authentication, too, and also the DynDNS domain configured in the Network settings or the Windows computer name.
• Internal registration interval: This defines the time after which Cactus requires a re-registration from IP phones. Default is 180 seconds (three minutes).
• External registration interval: Defines after which time Cactus will re-register itself at SIP providers. Default is 600 seconds (ten minutes).
• Only local registrations allowed: On by default, hindering external IP phones somewhere in the Internet from registering at your server. Must be switched off if external branches are allowed; a DynDNS domain or static IP is also required in this case.
• Force authentication for registrations from LAN: On by default. If disabled, local IP phones can register at the server even if they are not configured as branches. We recommend to leave this on, otherwise people having access to your building would be able to receive calls.
• Force authentication for calls from LAN: On by default. If disabled, local IP phones can make calls even if they are not configured as branches. We recommend to leave this on, otherwise people having access to your building would be able to abuse your server.
• Automatic external access: If enabled, Cactus will not require an escape digit in front of a number. The access digit 0 will be added automatically to numbers which are not in the user list and are longer than the maximum length of internal ISDN numbers. However, it is still possible to use an asterisk and a provider number to select a specific provider (or *0 for ISDN). Leave this disabled if your internal ISDN numbers have 4 digits or more so that a safe decision internal/external is impossible, if you require project numbers or if you need to distinguish business from private calls. This setting affects all telephones and is disabled by default.
• Country code: Enter your own country code here including 00 at the beginning, e.g. 0044 for UK or 001 for USA/Canada. If this value is 001, Cactus uses µ-Law as ISDN encoding, otherwise A-Law is used as required for Europe and other countries.
• Insert area code if required: If local numbers can be dialled without an area code at the telephones used with Cactus, the area code is added if the number does not start with 0 and if it is not a special number.
• Area code: Enter your own local area code here, e.g. 020 for London or 415 for San Francisco. Note that if your country code is 001, Cactus expects a 3-digit area code here, otherwise a leading 0 is required as usual in Europe.
• Special numbers not starting with 0: The area code will not be added to numbers (e.g. 112) which start with a digit sequence (e.g. 11) entered here. You may enter multiple sequences delimited by commas.
• Diversion for no answer after ... seconds: This is the time after which a call will be diverted if it is not accepted at the called branch. The default value is 20 s. (Each ringing sound is equivalent to about 4 seconds.)

STUN/router

An external STUN server (Simple traversal of UDP over NAT) is used by the Cactus server if the Internet access does not use Windows RAS but a NAT router (Network Address Translation) in order to find out its own public IP address:

• Server address: Domain name of the STUN server to be used, e.g. stun.sipgate.net. Note that this does not need to be the STUN server of the provider you are actually using (though it should be under normal circumstances).
• Server port: Port number of the STUN server. Default is 3478, but Sipgate uses 10000.
• Update period: Default is 180 seconds (three minutes). Should be 60 to 900 typically.
• NAT keepalive: If a port forwarding of port 5060 to the IP address of the Cactus server is configured in your router (recommended, if possible!), enter 0 here to avoid unnecessary traffic. Otherwise the value should be slightly below the NAT timeout of your router; default is 50 seconds. Cactus will then send empty UDP packets to all SIP providers using the given interval to keep the router port open. The setting is ignored if no STUN server is entered, which is typical if no router is used but RAS (PPPoE or similar).

If no router is used but Windows RAS (PPPoE driver or similar), the field for the server address should be empty to avoid unnecessary traffic. In this case, Cactus will retrieve the public IP address from the RAS driver once per minute.

If you have a static external IP address, enter it in the field "Server address" and leave the "Server port" empty.

ISDN

The ISDN configuration is required only if an ISDN adapter with a CAPI 2.0 driver is present.

• External access prefix: If the ISDN adapter is a branch within an ISDN PBX which requires an escape code for external access, enter this digit here (typically a 0). Leave this field empty otherwise.
• Active controllers: If you have more than one ISDN controller, you can select here which controllers are actually used for outgoing MSNs which are configured for "any" controller.
• Length of internal numbers: If the ISDN adapter is a branch within an ISDN PBX which requires a digit for external access, enter the maximum length of local branch numbers here (including any group numbers). Enter 0 here otherwise.
• Suppress own number: Your number will not be signalled for outgoing ISDN calls if this option is checked. The option requires the CLIR service enabled in the public exchange (Calling Line Identification Restriction), otherwise it does not have an effect.
• Tone dial timeout: If the caller "forgets" to dial # behind a number, the call will be initiated after the timeout configured here. Values from 4 to 99 are allowed, default is 5 (just as in many IP phones).

The following parameters are available for the entries in the MSN list (multiple subscriber number):

• ISDN controller: Enter 1 if you have a simple ISDN adapter with just one controller/bus. Otherwise enter the corresponding bus number for each MSN. (Note that this is not a channel number since one ISDN bus typically has two channels.)
• MSN: The "Multiple Subscriber Number" is one of the local numbers of an ISDN bus. Within an ISDN PBX, this is a (short) branch number. At an external ISDN line, it is typically one of your local ISDN numbers without the area code. It is not a problem to use the same ISDN adapter for more than one application, e.g. for a fax program and Cactus, but the MSNs must differ. If your ISDN PBX does not signal any MSN, you must enter a minus sign here.
• Use for outgoing call: MSN will be used for IP phones for outgoing ISDN calls. The following options apply:
  • No: Will never be used for outgoing ISDN calls (but can be used for incoming calls, see below).
  • Business: Outgoing external calls with 0 as escape code, but also for routing incoming SIP calls to the ISDN PBX.
  • Private: Outgoing external calls with 9 as escape code.
  • Special assignment, e.g. GSM box: This option is intended for using the given ISDN line for special area or country codes only as configured in the routing section, e.g. a ISDN-GSM gateway box for mobile number ranges. A possibly configured external access digit is ignored in this case.
• Incoming ISDN call: Allows incoming ISDN calls to be routed to IP branch numbers. The following options apply:
  • Ignore: Calls on this MSN will be ignored completely. Alternatively you can delete this number from the list.
  • VoIP destinations entered with DTMF: Cactus accepts the call and uses touch-tone digits for the destination.
  • Fixed VoIP destinations: Enter one or more IP numbers, use commas for separation. It is also possible to enter e.g. *212345 to route the ISDN call to 12345 using the SIP provider 2.
  • Number for outgoing VoIP calls (CLIR): If you enter a number here and if your SIP provider supports it, this number will be displayed at the called party instead of the one entered for the SIP provider (CLIR=calling line identification representation). Most providers require a number format like 44... for UK (starting with the country code, no leading 0). Leave this field empty if not required.

Please note the following:

1. Do not use external MSNs to Cactus MSNs in the ISDN PBX if they are configured as "VoIP destinations entered with DTMF". Alien users could abuse your server for expensive calls.
2. The number of MSNs does not directly correspond with the number of simultaneous calls. However, in some ISDN PBXs only one call is possible over one MSN. To avoid problems, Cactus uses MSNs alternatively if an external access digit is configured for ISDN. If not, only the first configured MSN will be used.
3. A special NT-mode ISDN controller (network termination) is displayed if present, but no configuration is required.

SIP providers

You can configure up to nine SIP providers which will used for external connections. When you click on [New], default parameters of some common providers will be inserted. We recommend not to create multiple accounts for using multiple numbers but - if the provider supports it - add numbers to an existing account; otherwise you will get in trouble if you need more than nine numbers.

• Index: Enter a number from 1 to 9. This can be used by phones with *1 to *9 in front of a destination number to select one SIP provider. 0 is reserved for ISDN
• Real name (screen name): Your VoIP phone number for the specified SIP provider. Note that this number must also be configured in the call distribution to specify which phones should ring for incoming calls.
• User-specified display name if applicable (CLIR): Enable this option only if this account supports individually different display numbers as set in the user configuration for the calling line identification representation.
• Account (username): The account name which your provider gave you for this account. Often this is a network-internal number.
• Authentication name (optional): Used by some providers; if empty, the username is used for authentication.
• Password: The password configured at your SIP provider.
• Realm: The SIP realm, mostly the same as in the registrar field.
• Registrar: Enter the domain name of the provider's SIP server (e.g. sipgate.net).
• Port: The SIP port number of your provider, typically 5060.
• Simultaneous channels: Some SIP providers allow only one log-in per account. If this is the case, enter 1, otherwise a higher value like 2 or 5. You can also enter 0 here to disable an account temporarily without having to delete its configuration.
• Use following line as fallback: You can select ISDN here or an alternative SIP provider which will be used if the current provider is not available or all of its channels are busy.

Users/Extensions

Here you can define the branch numbers, usernames and passwords of your Cactus PBX. No entries are required here if you only intend to use conventional telephones connected to an existing ISDN PBX.

Accounts

• Number: The branch number should have two to four digits typically. While alpha names would be possible technically, we recommend numeric branches to allow dialling with a normal telephone keypad. If you are using an ISDN PBX, too, please make sure that the first digit of the IP branches differs from the ISDN and analogue branch numbers like in the sample numbering plan on the right.
• Name: Used for writing names in the user list and in the PBX log. All characters except commas are allowed.
• Password: The password should have 4 to 8 characters and is case-sensitive. It is used for IP phone authentication. Digits are preferred for easy entry. Note that this is not the configuration password of IP phones but the registrar authentication.
• Outgoing provider: If you do not want to use the default routing for this branch, a special SIP account or ISDN MSN can be selected here. (Yet a different SIP provider can be used by dialling *1 to *9 in front of the number or *0 for ISDN. Also, a call distribution rule like 112 to *00112 would still work.)
• Number for outgoing VoIP calls (CLIR): If you enter a number here and if your provider supports it, this number will be displayed at the called party instead of the one entered for the SIP provider (CLIR=calling line identification representation). Most providers require a number format like 44... for UK (starting with the country code, no leading 0). Leave this field empty if not required. - Please make sure that a call distribution rule for incoming calls is also defined for every SIP number used here.
• One call only: If enabled, Cactus will send a busy signal if a call is routed to an engaged user. If disabled and if the IP phone supports multi-line operation, the user can switch between multiple connections.

Note: If the SIP server configuration does not require a password for authentication in the local-area network, an IP phone can register with any branch number even if it is not configured here (not recommended!).

Diversions

• Always/immediate: Calls are always and immediately diverted to the given destination number.
• If busy: Calls are diverted if this user is busy. This depends on the number of lines available in the phone and on the option "One call only" (see above).
• If not available: Calls are diverted if the phone is not registered at the server, e.g. because its power is switched off.
• If not answering: If the user does not accept the call within a given time frame (typically 20 seconds), the destination number will ring in addition. Effectively this diversion type implements a delayed parallel call, so the original branch can still accept the call.

Diversion notes:

1. The redirection is ignored if the calling party is identical to the diversion destination (secretary function).
2. If the destination is a mobile phone, please make sure with your network provider that the call is not connected unless it is really accepted by the called person; otherwise the other telephones in a calling group would no longer ring.
3. Disabling a diversion type does not clear the destination number field, which makes it easy to re-activate it later on.
4. Many SIP telephones have a "Do not disturb" button, or allow to reject a call. These functions will typically answer calls with a "Busy" status.
5. Some SIP telephones have diversion options of their own; however, they will not work for external callers or when the phone is switched off. This is why the Cactus diversion function should be preferred. For the same reason, the seconds for a time-based diversion should be set to a higher value in the telephone than in Cactus (e.g. Snom).
6. When using Cactus for diverting calls from and to external numbers, the caller pays for the call to Cactus only, and you pay for the outgoing connection from Cactus to the destination number. As a result, you will not get rich by diverting calls to a 0900 premium number owned by you, but poor.
7. Please take into account that an ISDN bus has 2 channels only! A call from a local ISDN phone on the same bus as Cactus cannot be diverted to an external ISDN number because this would require 3 channels.

Privileges

The following user privileges can be enabled or disabled selectively:

• Diversions programmable by phone: To avoid abuse, leave this option disabled if non-trusted persons have access to the user's telephone.
• National calls: National SIP and ISDN calls are allowed.
• International calls: International SIP and ISDN calls are allowed. If enabled, national calls are allowed, too.
• Manual provider selection: The user is allowed to prepend *0 to *9 to the destination number for a specific provider (0=ISDN, 1-9=SIP).
• Manual MSN selection: The user is allowed to prepend *MSN* to the destination number for using a specific MSN.
• Registration does not time out: If enabled, the SIP phone will stay registered even if it does not re-register itself every few minutes. The option should be used only if it is required due to firmware problems in the phone and if its IP address will never change.

Call distribution

The call distribution defines which phones should ring for incoming and outgoing calls or define speed-dial numbers. It is even possible for several phones to ring in parallel. When creating a new rule, the following fields are available:

• Called number: The number dialled originally, e.g. 112, 7654321 or 032987654321. You can also simply enter an asterisk * here, this rule will be used for all incoming external calls if no other call distribution rule is found.
• Description: A display text like "Fire brigade" or "My SIP number".
• Destinations: The number(s) which should be called for this entry, e.g. 0112 or 31,32,33. If an ISDN PBX is present, the destinations can also be ISDN or analogue phones. External destinations like cell phone numbers are possible as well. If more than one number is given, the first one accepting a call will get it.

The call distribution can als be used for speed dial numbers. For instance, you might create a short code 22 as an alias for *150000. This will select *1 for Sipgate as provider index 1 and 50000 for Sipgate's voice mailbox.

Please note:

1. The distribution of incoming ISDN calls to IP numbers is not defined here but in the ISDN configuration.
2. You must also enter the number(s) here which you got from your SIP provider(s). The local branch numbers which should ring are entered as destinations.
3. Short-access codes can be specified here, too, but please make sure that these do not conflict with existing internal or external numbers.
4. The call distribution is used before the routing rules. If you have a call distribution rule e.g. from 112 to 0112, you need 0112 (not 112) in the routing rule. You can also enter e.g. *0 in front of a destination number to override a default provider given in a user configuration by ISDN; in this sample, the destination would be *00112.
5. If an asterisk and a provider index is dialled (e.g. *1) for using a specific provider, the call distribution is normally ignored so that it is possible to dial user numbers of a given SIP provider. However, you might also enter an asterisk at the beginning of a called number in the call distribution to reroute it.
6. Complete numbers are required for the called number and all destinations (while the routing rules can use abbreviate numbers).
7. Cell phone numbers can be used as destinations. However, if conventional phones should ring at the same time, please make sure that the call is not accepted on the mobile number when you are out of range or if the cell phone is out of service.
8. For ringing several ISDN branches in parallel, define a group number in the ISDN PBX and configure it as one destination in Cactus to avoid wasting ISDN channels.
9. Call distribution rules are used before any diversions.

Nightly exceptions

It is possible to enter call distribution exceptions here which are valid during a specified time of day and optionally also 24 hours for Saturday and/or Sunday and national holidays, typically outside of your office hours. The form of these exceptions is the same as for the normal call distribution rules. If no matching rule is found within these, the normal distribution is used. - Some tips:

1. These rules override any diversions.
2. If you do not want to be disturbed during the night, simply leave the destination field empty, or enter the extension of an automatic answering machine. This could also be the ISDN number of a Shamrock CapiCall voice mailbox.
3. When you are on vacation, temporarily enter e.g. 08:00 to 07:59 as night time until you are back.
4. National holidays are configured in a file FEIERTAG.INI. If Shamrock CapiCall is installed on the same PC, the content of this file can be replaced by a single line pointing to the holiday list in the CapiCall folder, e.g. c:\capicall\feiertag.ini, so only one file has to be edited later.
5. Clicking the [Holidays] button on a different network workstation does work if you have started the configuration program from a mapped network drive, but not using a UNC path like \\server\c\...

Routing

The routing configuration defines which SIP/ISDN provider should be used for given country codes, area codes or special numbers. To be able to make external calls, at least a routing for the external escape code 0 must be defined here.

• Number prefix: The code including the external access digit, e.g. 0 for an external business call, 9 for an external private call, 00 for long-distance calls, 000 for international calls, 00033 for France and so on, 3 for internal branches beginning with 3, and so on.
• Description: Enter a display name for the code, e.g. External business, France, and so on.
• Routing: Select the desired routing, e.g. one of your SIP providers or ISDN. (You must configure the SIP providers first. An entry "ISDN special" will occur only if at least one MSN is configured as "Special assignment".)

If you are using an ISDN adapter connected as an internal branch of an ISDN PBX, enter rules for all possible ISDN branch numbers with "ISDN" as routing so that all ISDN branches can be called from IP telephones and SIP providers, too. For instance, if the telephones of the ISDN PBX are in the range 31 to 38 and 41 to 48 and group numbers are in the range 301 to 309, enter one rule from "3" to ISDN and also one from "4" to ISDN.

Please note:

1. The routing rules are not used if an asterisk and a provider number is used in front of the destination number. For instance, to dial the number 765 4321 of SIP provider 1, enter *1 765 4321 at the telephone.
2. The routing rules are not used either if a specific SIP account or ISDN MSN is preset in the user configuration.

Blocked numbers

You can block specific numbers or area codes for incoming and outgoing calls, e.g. to avoid expensive 0900 connections. Click on [New] to enter a number:

• Number: The first digits of the number range to be blocked (without external access digit), e.g. 0900.
• Description: A non-functional text, e.g. Premium services.

Important notes:

1. Many SIP providers block premium numbers themselves, e.g. 0900.
2. The specified numbers are blocked regardless of how the external access is dialled (0, 9, *0 to *9).
3. If your country allows call-by-call provider selection, make sure that you also block the call-by-call prefix codes to avoid that users can bypass the blocked numbers.

Log files

This dialogue allows to define what calls are written into the log. Default is OFF for all so that all calls are written:

• Only external connections: If enabled, calls between branches are not logged.
• Only outgoing connections: If enabled, incoming calls are not logged.
• Only successful connections: If enabled, failed calls (e.g. busy) are not logged.
• No private connections: If enabled, no calls with the private dial prefix 9 are logged.

Project numbers are frequently used to log the time used for specific customers or projects. The corresponding settings are available if the automatic external access (see SIP server) is disabled:

• Dial prefix: Enter the dial prefix for project numbers, e.g. 8. Please make sure that this does not conflict with branch numbers or other prefixes such as 0 or 9.
• Length: Enter the desired length of your project numbers here, e.g. 2 or 4.

For example, if you want to use the project number 12 for dialling an external business call to 02012345, enter this on your phone (assuming that the project number prefix is 8):
8 12 0 02012345
Note that the external access digit 0 is entered after the project number. This way even internal calls can contain a project number.

For analyzing possible problems, e.g. regarding the communication with a specific IP telephone or with a SIP provider, there is another option:

• Trace file SIP_TRACE.TXT enabled: Off by default. You should enable this option temporarily only since it creates an additional CPU load which is not useful under normal circumstances. SIP communication. About the last MByte is visible.

If the Cactus program is terminated abnormally (e.g. power outage, PC with Cactus as a service is shut down, or crash), an existing file SIP_TRACE.TXT will be renamed to SIP_TRACE_PREV.TXT during the next restart. If SIP_TRACE_PREV.TXT already exists in this situation, it will be overwritten.

Program start

Cactus can be launched as a Windows service so that is available even if no user is logged in. We recommend to do this only after your configuration is stable and everything is working since newer Windows versions (Vista and later) do not allow to display a service as a symbol in the system tray or to open it as a window; the activities can only be seen in the log files. Furthermore some configuration changes (like SIP port) require a restart. - These buttons control the operation:

• Install Cactus as a Windows service: Quits the Cactus server if it is still running as a normal application, creates an entry "Cactus-PBX" in the Windows services list, and starts this service.
• Stop Cactus Service: Stops the service and removes "Cactus-PBX" from the Windows services list.

If Cactus is launched from a network drive (e.g. NAS device) or in a remote session, you will have to register the service manually. Open a command line as administrator and enter SIP_PBX.exe -i in the Cactus program folder to register the service. Then open the Services applet in the Windows Control Panel and start the Cactus service. Let the desktop option enabled so that the Cactus symbol is available in the system tray for opening the server window. If you want to unregister the service later, enter SIP_PBX -u in an administrator command line window.

Please note: When run as a service, Cactus must not be configured as a Startup link and cannot be controlled by the AppDog watchdog utility from Shamrock either.

On the same configuration page there is also a Language selection box (English or German). The selected language not only applies to the configuration program itself but also to the Cactus server, to the log viewer, and to any other multi-lingual programs from Shamrock Software on the same PC.

The Cactus Server

Starting the server

Please note that it may take a few minutes after starting the server until the IP telephones have registered and you can use them for calls.

Startup link, AppDog, service

We recommend to launch the Cactus server automatically using a Startup link to SIP_PBX.exe in the Windows menu Start/Programs. Alternatively, you can also use the free AppDog utility from Shamrock to launch and monitor the Cactus server. When your configuration is complete and stable, you can run the Cactus server as a Windows service so that it is available even with no user logged in.

Important: During your initial tests the server should not run as a service because some configuraton changes require a restart which is more cumbersome for a service.

The server window

The server window shows the status of SIP provider registrations, currently registered IP telephones, current connections and other relevant information. (In all current Windows versions it is only visible if Cactus is run as a normal application and not as a service!)

• To show the Cactus window or to minimize it, left-click on the Cactus symbol in the system tray section of the Windows task bar (near the time display).
• Clicking on [x] in the top right corner of the server windows does not end the program but minimizes it instead to avoid that all current calls are disconnected undeliberately.
• To end the server (if it does not run as a service) right-click on the Cactus symbol in the system tray; a context menu will appear allowing to stop the server. Alternatively the "File" menu of the server window can be used. If the server is restarted within the registration timeout, the registration status of the phones will not change (it is saved temporarily in phone.sav).

The window panes use the following colours for displaying status information:

• SIP providers: provider status, red = registration failed, green = registration succeeded.
• Users: SIP phone status, black = registered, green = phone freshly registered within the last seconds, grey = not re-registered but registration timeout is disabled.
• Actions: Line in black = general information, blue = SIP calls, magenta = ISDN calls, red = errors.
• Trace: Line in black = SIP reception, blue = SIP transmission, magenta = ISDN, green = general information.

The File menu of the server window contains these items:

• Configuration: Shows the configuration program.
• Debug on/off: Enlarges the server window so that debug information becomes visible. This information is the same as in the optional trace file SIP_TRACE.TXT.

The Help menu allows to display this manual and also the current software version.

Viewing the call log

SIP_LOG.exe can be launched on the server of from other workstations as well if the server hard disk is shared in the network. It displays a decoded version of the log file. You can change the window size by dragging the border with the mouse. The following columns are displayed:

• Date, time: The end of the call or connection.
• Duration: The duration of the connection (if successful).
• Ring: Shows how long the telephone was ringing before the call was answered or dropped.
• Caller: The caller's number (without the external access digit).
• Type: Source and destination, controller number (if more than one bus) and MSN for ISDN.
• Ringing: The called number(s), may be several for a group call.
• Connected: The number which actually accepted the call.
• Info: Codecs used in this connection or error code if the call failed.
• Project: This column is shown if project numbers are enabled in the configuration.

The information in the "Flags" field of the log file is used to display the lines in different colours:

• Green: Incoming successful external call
• Blue: Outgoing successful external call
• Black: Successful internal call
• Red: The call failed, see the Info column for the cause

The log file format

Calls handled by the Cactus server are logged in a monthly file PBXyymm.LOG with CSV-formatted lines (comma-separated values, yy=year, mm=month, e.g. PBX0702.LOG). You can configure which calls are logged. The file can also be read by other programs like Microsoft Excel. If you develop software on your own, please make sure that your read access does not block the write access from the Cactus server.

Line format

Each line uses the following format with commas between all fields:
Date, Time, Duration, Ring, Flags, Caller, Type, Ringing, Connected, Info, Project number
The parameters have the following meaning:

1. Date: The end of the call or connection: dd.mm.yy.
2. Time: The end of the call or connection: hh:mm:ss.
3. Duration: The duration of the connection in seconds (if successful).
4. Ring: Shows how long the telephone was ringing before the call was answered or dropped.
5. Flags: O=outbound, i=inbound, P=private, m=manual provider selection.
6. Caller: The calling number as transmitted by the provider.
7. Type: Source>destination; bus:MSN or "NT" for ISDN, "i" for internal branch, or number@registrar.
8. Ringing: The called number(s), may be several for a group call, delimited by semicolons.
9. Connected: The number eventually accepting the call.
10. Info: Either a 3-digit SIP error code, or a 4-digit ISDN CAPI error, or the codecs at Caller>Called, e.g.
    0=G.711A, 2=G.726, 3=GSM, 8=G.711µ, 97=iLBC, 110=Speex.
11. Project number: The project number used for this call if one has been entered.

Please note the following conventions:

• External numbers are displayed without an external access digit. The "Type" and "Flags" fields determine if it was an internal, private, or business call.
• The "Ringing" field shows only currently registered users. As a result, it is possible that calls are shown without any ringing number.
• For ISDN connections, the type field contains the bus (controller) number, a colon and the MSN, e.g. i>2:43.
• SIP calls typically use an international number format even for local calls; this is also written into the log file.

Here is a sample of an outgoing external call lasting 62.7 minutes and 1.7 seconds where the branch 52 calls the ISDN number 08912345 using the MSN 43 on bus 1 with the project number 4711:
25.03.2007,11:59:15,62:01.7,12.8,O,52,i>1:43,08912345,08912345,8>8,4711

The following line shows an incoming call from 08912345 on ISDN bus 3, MSN 48. The IP branches 52 and 57 ring for three seconds until 52 accepts the call:
25.03.2007,11:51:02,1:12.2,3.0,I,08912345,3:48>i,52;57,52,8>8,

Error codes

The "Info" field of the log file can contain 3-digit SIP or 4-digit ISDN/CAPI error codes. If several phones are called as a group, only the last error is logged. The following table shows the most typical codes.

The server log

SIP_PBX.TXT is another log file written by the Cactus server. It contains all server-relevant events like program start/end, public IP detection, and problems with registrations. This log will be deleted automatically when it reaches a size of 100 KB. It can be viewed in the File menu of the call log viewer.

The server log also shows if there are new messages in the voice mailbox of a SIP provider. To call the mailbox from a telephone, a short code in the call distribution list is useful.

Repeatedly occurring errors, new voice messages at SIP providers and incoming calls can also be displayed on other LAN workstations if they run the Popup program which comes with Shamrock's NetMail software. If Cactus is installed on the NetMail gateway PC, it uses the popup destination address for system messages in the gateway configuration, otherwise the messages are sent to * (all). In addition, the popup program also imposes a status box showing all currently active calls.

Using Phones

Configuring IP phones

You must configure your IP phones properly before they can be used. Most phones have a web interface. Typically the following settings are available:

• Registrar: For phones within your local-area network, enter the IP address of the Cactus server, or the computer name of the server if it can be resolved by a local DNS server. For phones used as external branches of your server enter the DynDNS domain of your server, e.g. myserver.dyndns.org. Alternatively you can enter the static address of your server if you have one.
• SIP port: Normally 5060. If an external IP telephone cannot register at the server, e.g. because a router with built-in VoIP functions blocks this port, enter the port 2 configured in Cactus, typically 5062.
• Realm: If the telephone has such a setting, enter the same name as in the realm field of the Cactus SIP server configuration.
• DNS server: Not required if a IP address is used as registrar, otherwise either your local DNS proxy or the DNS IP address of your Internet provider.
• Username and Password: The username is identical with the branch number, e.g. 57. Username and password must match the user data configured in Cactus.
• Codecs: Most IP phones offer several codecs with different bit rates and speech quality. Please make sure that at least one of the two variants G.711a and/or G.711µ is enabled (sometimes called PCMA and PCMU) since Cactus needs them for its built-in ISDN gateway.

Using IP telephones

Please contact the manual of your IP phone(s) how to dial a number. Often the # key must be pressed after the number.

Incoming and outgoing calls

• Incoming VoIP calls can be routed to one or more phones using the call distribution. The calling number is displayed including an external access digit.
• To call another branch, i.e. a different IP phone registered at your server or a phone connected to a local ISDN PBX, simply enter the branch number.
• To call an external number, use 0 as prefix for business calls or 9 for private calls. This prefix is obsolete if an automatic external access is configured.
• To call an internal number at a SIP provider, e.g. at SipGate, press the asterisk key followed by the index as defined in the provider configuration and then the user number.
• To call an external number via a specific provider, press the asterisk key followed by the index as defined in the provider configuration and then the normal number including the area code. The country code is required for international calls.
• To use a specific ISDN MSN, dial *MSN*Destination, e.g. *42*08912345678. The MSN must not be 0. A prerequisite for this function is that the selected MSN is listed in the Cactus ISDN configuration; if there is more than one controller and the MSN is not valid on all controllers, a specific controller number must be assigned for the MSN.
• Transferring a call requires an IP telephone with a Flash and a Transfer key. Not all SIP phones are capable of transferring calls. Please have a look at the manual. Cactus uses the attended transfer mechanism as standardized in RFC 3515: A talks with B, then B wants to transfer the call to C. So B presses the flash key and dials C. When C accepts the call, B talks with C first. When B hangs up, A is connected to C.

Programming diversions

If programming diversions at the telephone is allowed, the following key sequences can be used for it which are similar to the ones being standardized for GSM cell phones:

• **21*Number#: Always/immediate - all calls are immediately redirected to the given number.
• **62*Number#: Not available - diverts calls if the telephone is currently not registered at the server.
• **61*Number#: Not answering - diverts calls if they are not accepted within ~20 seconds.
• **67*Number#: Busy - diverts calls if the user is talking and has no free lines.
• **002*Number#: All - Sets all above diversions to the given number.
• **004*Number#: Conditional - Sets all diversions except "always/immediate".

Important notes for programming diversions:

1. After entering the sequence you have to lift the headset or press the dial key on the phone. The pound/hash key # finishes the entry for some telephones (e.g. Grandstream BT100) and is not transmitted itself. It is therefore optional if the telephone uses a different key for dialling. Consequently **002 is sufficient for disabling all diversions, for instance.
2. After a successful programming the telephone shows either "487" (if it can display status codes only) or "487 Diversion OK". Otherwise an error code is displayed depending on the cause, typically 403 = forbidden or 400 = bad format.
3. To disable the diversion, simply leave the Number field empty (e.g. **21*# or even shorter **21#). You can see the current status in the user configuration.
4. A user who is the destination of a diversion can call the redirected telephone without being diverted again (secretary function).

Using an ISDN PBX

If the Cactus PC is equipped with an ISDN adapter, it can be used as a VoIP/ISDN gateway. Telephones connected to a conventional ISDN PBX can then be used for incoming and outgoing SIP calls.

• Incoming SIP calls can be routed to telephones connected to the ISDN PBX. The first digit(s) of these telephones must be entered in the call distribution, and they must have ISDN as routing destination. The displayed caller number will always be the MSN of the ISDN adapter since it is technically impossible to override it.
• To call an IP phone registered at the Cactus server, dial an MSN of the ISDN adapter first which is configured for DTMF. You will hear a dial tone. Now enter the desired branch number and press # at the end.
• To call an external number using SIP, dial an MSN of the ISDN adapter first which is configured for DTMF. You will hear a dial tone. Now enter the desired external number including the external prefix 0 or 9 (if no automatic external access is configured) and press # at the end. Be careful not to make a pause longer than five seconds. If you mistype the number, use the asterisk key to re-enter it.
• To call an internal number at a SIP provider, dial an MSN of the ISDN adapter first which is configured for DTMF. You will hear a dial tone. Now enter an asterisk, the provider index and the user number to be called.
• To call an external number using a specific SIP provider, dial an MSN of the ISDN adapter first which is configured for DTMF. You will hear a dial tone. Now enter an asterisk, the provider index and the number to be called. You must include the area code even for local calls. For international calls, the country code is required.
• Transferring a call from an ISDN or analogue phone is possible, too. Please consult your ISDN PBX manual.

More hints for using analogue and ISDN phones:

1. It is useful to program a short-access key on your phone for dialling a Cactus ISDN MSN.
2. It is not mandatory to press # behind the destination number. You can simply wait a few seconds alternatively. The timeout can be programmed in the ISDN settings of Cactus.
3. If an extension of an ISDN PBX rings during an incoming SIP call, not the SIP caller number is displayed but the internal MSN of the ISDN card used by Cactus. It is technically impossible to override this.
4. Programming diversions (e.g. on busy) is usually done at the local ISDN PBX.

Multiple Cactus PBXs

You can link multiple Cactus servers so that their branches can dial each other directly. To keep things simple, the first digits of the branch numbers of the different locations should differ. Furthermore, you will either need a static IP address or a DynDNS domain for each location.

You must buy a Cactus license for each location since one license is valid for one server only. - The following example assumes a configuration with two locations. Please follow these steps:

Now e.g. branch 51 at Cactus 1 can simply dial 61 even though it is at Cactus 2. - Notes:

1. If an ISDN PBX is available at both locations and if there are conflicts with their branch numbers (e.g. 31 exists at both locations), you can create a call distribution rule e.g. from 531 to 31 at location 1, and from 631 to 531 at location 2.
2. Even if theoretically possible, you should not use Cactus 2 for external calls from Cactus 1 and vice versa. It is much more efficient to have different SIP provider accounts at both locations.

Typical problems

Outbound calls do not work.
Please make sure that you have configured your own country code correctly, e.g. 0044 for UK. Also note that you have to dial e.g. 00033... instead of 0033 for France: An additional 0 is required for external calls if you did not enable the automatic external access. Otherwise Cactus would think that you have dialled a national number and adds the international prefix. If you have configured a branch-specific CLIR number, it must be supported by the provider; check if it works if this field is empty.

No phone is ringing when a call comes in.
Please check what is displayed in the upper right corner of the Cactus window. If "(?)" is shown behind the called number, no call distribution rule was found. Please create a rule for the displayed number, or enter an asterisk * as a wildcard for the called number. If a phone connected to an ISDN PBX should ring, please make sure that its first digit exists in the routing and at least one ISDN MSN is configured for outgoing business calls.

I hear an annoying echo.
Since IP telephony uses data buffers, a speech delay of a fraction of a second cannot be avoided. If microphone and speaker are not decoupled well, this may result in an echo. Often reducing the volume or the microphone sensitivity helps. But it is always best to use a headset with softphones, or at least a headphone instead of a speaker. If a conventional ISDN or SIP telephone is used, the speakerphone mode should be enabled only if the phone has a built-in echo cancellation. In general, ISDN and SIP phones cause less echo problems than analogue ones.

I cannot dial a number with an asterisk in front of it.
Some ISDN PBXs require extensions to be dialled with an asterisk * in front of them. But Cactus interprets an asterisk in the first place as a provider selection, and two asterisks are reserved for programming diversions. So instead of dialling e.g. *31 for the extension 31, please enter *0*31 at your SIP phone: *0 selects ISDN as outgoing line, and the following *31 is then transferred to the ISDN PBX. An alternative might be to disable the automatic external access in your ISDN PBX for the MSNs used by Cactus. - Similarly it is possible to dial a sequence starting with * at a SIP provider if you prepend it with an asterisk and the provider number.

Can I use a GSM gateway box?
A GSM-ISDN gateway can be used with Cactus so that calls to and from mobile numbers are much cheapter because they are network-internal calls. You will need a GSM box which behaves like an ISDN NT for this purpose, e.g. a 2N BRI Lite. Connect it to one reserved outlet of the ISDN card in the Cactus PC and define its MSN and controller number as "Special assignment" in the ISDN settings. Then configure a proper routing for the mobile area code(s) to "ISDN special".

License contract

The owner of a Cactus license is authorized to install and run the server software on one PC. The other modules may be started from any other PC within a local network if they are loaded from the same server PC.

Distribution of the demo version on other web sites than the ones owned by Shamrock or on CD ROMs is prohibited. Shamrock Software is the author of the program and of this manual and reserves all rights, including copyright regulations.

Shamrock Software grants the right to use the software but does not transfer ownership. This right ends if the invoice is not paid in time or if any term of this license agreement is violated by the user.

You may not use this software in any circumstances where there is any risk that failure of this software might result in a physical injury or loss of life. Any such use is entirely at your own risk, and you agree to indemnify Shamrock Software from any claims relating to such unauthorized use.

© Shamrock Software GmbH