3CX Backup & Restore Commands

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Introduction
Starting from v15.5 SP1, backup and restore command line tools are included with the 3CX PBX on Windows and Linux. These commands can be used to create scripts and schedule backup and restore operations, externally to the built-in 3CX Management Console functionality. This is useful when hosting on cloud for archiving in bulk PBX users, data and configuration for safekeeping.


BackupCmd
The BackupCmd command line tool enables backups to be taken with these options:

-l, --log=VALUE

Log filename or full file path, e.g.

 

--log=/var/lib/3cxpbx/bak_cmd_run.log
-f, --file=VALUE

*Filename to backup in default backup location or full zip archive path, e.g.

 

--file=/home/pi/backups/backup.zip
-o, --options=VALUE

Backup options, specifying to include in the backup:

ALL - include everything, or
add following comma-separated options to only include:
CH - Call History
LIC - License
FQDN - Phone System FQDN
PROMPTS - Voice Prompts
FW - Phone Firmware
REC - Recordings
VM - Voicemails
--pwd=Value

Encrypt backup files with password (V15.5 Sp2 onwards), e.g.

 

--pwd=My_B@ckup_Pa$$
-h, --help

Display command help

Notes:

Mandatory options are denoted above with *.
Specifying an existing backup file with the -f or --file switch, overwrites it without warning.
Specify an existing log file with the -l or --log switch to append to.
Values, filenames and paths with space(s) are not recommended, otherwise they need to be enclosed in double quotes when used in option arguments.
Ensure that specified backup and log file paths are writable on Linux by the “phonesystem” user, and on Windows by the executing user.


General
Specifying a filename with the

--file or --f switch as a:
filename, the backup file is stored in the local, FTP, or Google Drive directory configured as the backup location in Management Console, e.g.

-f=my-pbx_full_bak.zip
full filename path, overrides the configured location and stores the backup file in the specified path, e.g.

-f=c:\backup\my-pbx_full_bak.zip
When the

--log argument is specified, it has to be followed by either:
a filename to store the log in the current working directory, e.g.

-l=bak_cmd.log -f=my-pbx_full_bak
a full path to a local filename to override and store in the file specified, e.g.

-l=c:\backup\bak_cmd.log -f=my-pbx_full_bak



Backup Command on Windows
To use the backup command on Windows, open a command prompt with administrative privileges, change to this directory using

cd C:\Program Files\3CX Phone System\Bin and run the command to:
Display available backup command options:

BackupCmd.exe --help
Make a full PBX backup and keep a log:

BackupCmd.exe --file=full_pbx_backup.zip --options=ALL --log=backup_cmd.log
Make a backup including call history, license and FQDN, keeping a log:

BackupCmd.exe --file=partial_pbx_backup.zip --options=CH,LIC,FQDN --log=backup_cmd.log



Backup Command on Linux
To use the backup command on Linux, run the command in a terminal as user

phonesystem using sudo to:
Display available backup command options:

“sudo -u phonesystem 3CXBackupCmd --help” command to see all available options.
Make a full PBX backup and keep a log:

/var/tmp/pbx-backup_cmd.log
Make a backup including call history, license and FQDN, keeping a log:

/var/tmp/pbx_backup_cmd.log



RestoreCmd
The RestoreCMD tool enables to restore backups via command line with these options:

-l, --log=VALUE

Log path or filename

-f, --file=VALUE

*Backup path or filename to restore

-h, --help

Show command help

--pwd=Value

Decrypt backup with given password (V15.5 Sp2 onwards)

--failover

Failover mode - services are not started after restore on a PBX set up as passive failover node

Notes:

Mandatory options are denoted above with *.
Specify an existing log file with the -l or --log switch to append to.
Values, filenames and paths with space(s) are not recommended, otherwise they need to be enclosed in double quotes when used in option arguments.
Ensure that specified log file paths are writable on Linux by the phonesystem user, and on Windows by the executing user.


General
Specifying a filename with the

--file or --f switch as a:
filename, retrieves the backup file from the local, FTP, or Google Drive directory configured as the backup location in Management Console, e.g.

-f=my-pbx_full_bak.zip -l=c:\backup\restore_cmd.log[/code]full filename path, overrides the configured location and retrieves the backup file from the specified path, e.g.RestoreCmd -f=c:\backup\my-pbx_full_bak.zip -l=c:\backup\restore_cmd.log
When the

--log argument is specified, it has to be followed by either:
a filename to store the log in the current working directory, e.g.

-l=bak_cmd.log -f=my-pbx_full_bak
a full path to a local filename to override and store in the file specified, e.g.

-l=c:\backup\bak_cmd.log -f=my-pbx_full_bak
Note that it needs to be writable path for “phonesystem” user on Linux and on windows based on the user the schedule is been executed with..


Restore Command on Windows
To use the restore command on Windows, open a command prompt with administrative privileges, change to this directory using

cd C:\Program Files\3CX Phone System\Bin and run the command to:
Display available restore command options:

RestoreCmd.exe --help
Restore a backup and start 3CX services immediately after restore:
RestoreCmd.exe --file=full_pbx_backup.zip --log=restore_cmd.log
Restore a backup in failover mode and keep 3CX services stopped on a PBX set up as passive failover node:
RestoreCmd.exe --file=full_pbx_backup.zip --log=restore_cmd.log --failover



Restore Command on Linux
To use the restore command on Linux, run the command in a terminal as user

phonesystem using sudo to:
Display available restore command options:
sudo -u phonesystem 3CXRestoreCmd --help command to see all available options.
Restore a backup and start 3CX services immediately after restore:
sudo -u phonesystem 3CXRestoreCmd --file=full_pbx_backup.zip --log=restore_cmd.log
Restore a backup in failover mode and keep 3CX services stopped on a PBX set up as passive failover node:
sudo -u phonesystem 3CXRestoreCmd --file=full_pbx_backup.zip --log=restore_cmd.log --failover

Can You List All Known SIP Responses?

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Can You List All Known SIP Responses?
SIP responses are the codes used by Session Initiation Protocol for communication. We have put together a list of all the SIP responses known.

1xx = Informational SIP Responses
100 Trying – Extended search is being performed so a forking proxy must send a 100 Trying response.
180 Ringing – The Destination User Agent has received the INVITE message and is alerting the user of call.
181 Call Is Being Forwarded – Optional, send by Server to indicate a call is being forwarded.
182 Queued – Destination was temporarily unavailable, the server has queued the call until the destination is available.
183 Session Progress – This response may be used to send extra information for a call which is still being set up.
199 Early Dialog Terminated – Send by the User Agent Server to indicate that an early dialogue has been terminated.
2xx = Success Responses
200 OK – Shows that the request was successful
202 accepted – Indicates that the request has been accepted for processing, mainly used for referrals.
204 No Notification – Indicates that the request was successful but no response will be received.
3xx = Redirection Responses
300 Multiple Choices – The address resolved to one of several options for the user or client to choose between.
301 Moved Permanently – The original Request URI is no longer valid, the new address is given in the Contact header.
302 Moved Temporarily – The client should try at the address in the Contact field.
305 Use Proxy – The Contact field details a proxy that must be used to access the requested destination.
380 Alternative Service – The call failed, but alternatives are detailed in the message body.
4xx = Request Failures
400 Bad Request – The request could not be understood due to malformed syntax.
401 Unauthorized – The request requires user authentication. This response is issued by UASs and registrars.
402 Payment Required – (Reserved for future use).
403 Forbidden – The server understood the request, but is refusing to fulfil it.
404 Not Found – The server has definitive information that the user does not exist at the (User not found).
405 Method Not Allowed – The method specified in the Request-Line is understood, but not allowed.
406 Not Acceptable – The resource is only capable of generating responses with unacceptable content.
407 Proxy Authentication Required – The request requires user authentication.
408 Request Timeout – Couldn’t find the user in time.
409 Conflict – User already registered (deprecated)
410 Gone – The user existed once but is not available here any more.
411 Length Required – The server will not accept the request without a valid content length (deprecated).
412 Conditional Request Failed – The given precondition has not been met.
413 Request Entity Too Large – Request body too large.
414 Request URI Too Long – Server refuses to service the request, the Req-URI is longer than the server can interpret.
415 Unsupported Media Type – Request body is in a non-supported format.
416 Unsupported URI Scheme – Request-URI is unknown to the server.
417 Uknown Resource-Priority – There was a resource-priority option tag, but no Resource-Priority header.
420 Bad Extension – Bad SIP Protocol Extension used, not understood by the server.
421 Extension Required – The server needs a specific extension not listed in the Supported header.
422 Session Interval Too Small – The request contains a Session-Expires header field with a duration below the minimum.
423 Interval Too Brief – Expiration time of the resource is too short.
424 Bad Location Information – The request’s location content was malformed or otherwise unsatisfactory.
428 Use Identity Header – The server policy requires an Identity header, and one has not been provided.
429 Provide Referrer Identity – The server did not receive a valid Referred-By token on the request.
430 Flow Failed – A specific flow to a user agent has failed, although other flows may succeed.
433 Anonymity Disallowed – The request has been rejected because it was anonymous.
436 Bad Identity Info – The request has an Identity-Info header and the URI scheme contained cannot be de-referenced.
437 Unsupported Certificate – The server was unable to validate a certificate for the domain that signed the request.
438 Invalid Identity Header – Server obtained a valid certificate used to sign a request, was unable to verify the signature.
439 First Hop Lacks Outbound Support – The first outbound proxy doesn’t support the “outbound” feature.
440 Max-Breadth Exceeded – If a SIP proxy determined a response context had insufficient Incoming Max-Breadth to carry out a desired parallel fork, and the proxy is unwilling/unable to compensate by forking serially or sending a redirect, that proxy MUST return a 440 response. A client receiving a 440 response can infer that its request did not reach all possible destinations.
469 Bad Info Package – If a SIP UA receives an INFO request associated with an Info Package that the UA has not indicated willingness to receive, the UA MUST send a 469 response, which contains a Recv-Info header field with Info Packages for which UA is willing to receive INFO requests.
470 Consent Needed – The source of the request did not have the permission of the recipient to make such a request.
480 Temporarily Unavailable – Callee currently unavailable.
481 Call/Transaction Does Not Exist – Server received a request that does not match any dialogue or transaction.
482 Loop Detected – Server has detected a loop.
483 Too Many Hops – Max-Forwards header has reached the value ‘0’.
484 Address Incomplete – Request-URI incomplete.
485 Ambiguous – Request-URI is ambiguous.
486 Busy Here – Callee is busy.
487 Request Terminated – Request has terminated by bye or cancel.
488 Not Acceptable Here – Some aspects of the session description of the Request-URI are not acceptable.
489 Bad Event – The server did not understand an event package specified in an Event header field.
491 Request Pending – Server has some pending request from the same dialogue.
493 Undecipherable – UndecipherableRequest contains an encrypted MIME body, which recipient can not decrypt.
494 Security Agreement Required – The server has received a request that requires a negotiated security mechanism.
5xx = Server Errors
500 Server Internal Error – The server could not fulfill the request due to some unexpected condition.
501 Not Implemented – The SIP request method is not implemented here.
502 Bad Gateway – The server, received an invalid response from a downstream server while trying to fulfill a request.
503 Service Unavailable – The server is in maintenance or is temporarily overloaded and cannot process the request.
504 Server Time-out – The server tried to access another server while trying to process a request, no timely response.
505 Version Not Supported – The SIP protocol version in the request is not supported by the server.
513 Message Too Large – The request message length is longer than the server can process.
555 Push Notification Service Not Supported – The server does not support the push notification serviced specified in the pn-provider SIP URI parameter.
580 Precondition Failure – The server is unable or unwilling to meet some constraints specified in the offer.
6xx = Global Failures
600 Busy Everywhere – All possible destinations are busy.
603 Decline – Destination cannot/doen’t wish to participate in the call, no alternative destinations.
604 Does Not Exist Anywhere – The server has authoritative information that the requested user does not exist anywhere.
606 Not Acceptable – The user’s agent was contacted successfully but some aspects of the session description were not acceptable.
607 Unwanted – The called party did not want his call from the calling party. Future attempts from the calling party are likely to be similarly rejected.

کدهای سیستم تلفنی

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Introduction
Dial codes are key/number combinations used to access functions within the phone system directly from your phone. The administrator can change these from the “Management Console > Settings > Dial codes” tab. This section will describe the default dial-codes.

Parking
If you wish to “park” a call and then pick up the call from another extension, you can do so by parking the call in the “Parking orbit”. Alternatively, you can use the “Shared Parking Orbit”. The differences are outlined here: https://www.3cx.com/blog/docs/call-parking/

To park a call
On an established call to your extension start the “blind transfer procedure” and transfer the call to *0[0-9]. For example, *01 will place the call in the parking orbit 1, *02 will place the call in the parking orbit 2, etc.

To pick up a parked call
Dial *10 to *19 where the 0 – 9 is the park orbit number the call was parked in. For example, calling *11 will pick up any calls parked in parking orbit 1, *12 will pick up any calls parked in parking orbit 2, etc.

Parking Multiple Calls
Parking orbits support parking of multiple calls in the same orbit. Therefore, when unparking, you can add the extension number from which the call was parked to be sure that you un-park the correct call.

For example, if extension 100 parked a call in park 0, this call can be picked up by another extension by keying in *10100. Omitting the extension number otherwise un-parks the longest parked call in the particular orbit.

Note: Call Pickup and Call Park are only available in commercial editions of 3CX Phone System

Pickup a call
If you hear a particular extension ringing, and you know that the owner of that extension is not able to take the call, you can take the call by using the “Call Pickup” feature. To do this, type:

*20*

This will redirect the active call from the specified extension to your extension. You can also retrieve a call from a ringing extension without specifying the extension number. To do this, type *20* followed by the send key. This will redirect the longest ringing call to your extension.

Change Profile Status
You can change your status by using the dial code in the following manner: *3[0-4]. 0 is “Available”, 1 is “Away”, 2 is “Do not Disturb”, 3 is “Custom 1” and 4 is “Custom 2”.

For example, dialing *31 from extension 100, will change the profile status of extension 100 to Away.

Connect to Voicemail of extension
To leave a message in the Voicemail box of a particular Extension: Dial *4.

For example, *4100, will leave a voicemail message in the voicemail box of extension 100.

Setting Do Not Disturb ON / OFF (Deprecating)
To specify DND (Do Not Disturb) for a particular extension:

Dial *60 to set the extension to Do Not Disturb: OFF
Dial *61 to set the extension to Do Not Disturb: ON
Log extension IN/OUT of queues
To login or logout an extension from Queues you can use the following dial codes:

Dial *62 to log extensions into the queues
Dial *63 to log extensions out of the queues
For more information on how to manage queue agent status refer to this guide: https://www.3cx.com/docs/pbx-queue-status/

Paging
The intercom feature allows you to make an announcement to another extension without requiring the other party to pick up the handset. The message will be played via the other phone’s speaker. The audio is two way, and the called party can respond immediately without picking up the handset. Paging is also a group right which must be assigned to an extension. The extension can then use paging within its extension group. If the right is missing, paging calls will be converted to regular calls.

Start an Intercom Call
Prefix the extension you wish to call with *9, followed by the extension. For example, to make an intercom call to extension 100 you should dial: *9100

Important: Intercom dial code is disabled by default. If you want your phone to Auto-answer (only applicable for legacy devices) you need to configure a dial code in the PBX Settings → Advanced → Dial Codes tab. This Dial code must be unique and must not conflict with any of the other dial codes.

Billing Code
The Billing Code allows you to tag specific calls with Billing codes in order to produce reports. So for example, you can have agents use different billing codes when dialing out in order to check total billing for agents or if you are making a call for a particular customer and you will bill them afterwards. To tag calls with billing codes you will need to use ** (default value). This allows you to add a tag to a call you want to make.

For example, whenever you make a call for a particular customer, you want to tag the call with that customer’s billing code (for example 3265), so that you can bill them. When making a call related to a particular customer, the caller is required to dial the number in the following format:

Destination-Number**3265.

For example, if the number is 17771231233, then the caller needs to dial 17771231233**3265

This billing code can be used as a filter in the 3CX Reports (and in particular the Call Report with filter to destination : Match Billing Code) to see how many calls were made using the specific billing code as shown below.

 

Here we see that the billing code has been entered into the “Match Bill Code” textbox. This will query and display all the calls that were made using the “3265” billing code tag.

 

Force 3CX IN or OUT of Office – Emergency Code
This dial code (which is unspecified by default for security purposes) is a code that when used, will set the entire PBX to either:

In office hours
Out of Office hours
Back to auto switching based on time
This code is designed for emergency services when you need to change the routes of all your VoIP Lines to IN or Out of office quickly with a single phone call.

Usage of this Dial Code
Let us assume that the administrator configures this dial code to be *64 and also assume that the global office hours are set from 9am till 5.30pm.

Setting 3CX to In Office Hours
If the Administrator dials *641 (appends a 1 to the code), a prompt will be played stating that “3CX is now set to IN OFFICE HOURS”. This means that the PBX will disregard the current time, or day (even if it is a holiday) and set all the routes of the Gateways, VoIP Providers and DID/DDI’s to route to the In Office hours destination.

Setting 3CX to Out of Office Hours
If the Administrator dials *642 (appends a 2 to the code), a prompt will be played stating that “3CX is now set to OUT OF OFFICE HOURS”. This means that the PBX will disregard the current time, or day (even if it is 10am for example) and set all the routes of PSTN Gateways, VoIP Providers and DID/DDI’s to route to the Out of Office hours destination.

Setting 3CX to Default Office Hours Operation
If the Administrator dials *64 (with nothing appended), a prompt will be played stating that “3CX is now using the DEFAULT OFFICE HOURS”. This means that in this example, the PBX will override the previous 2 commands and behave as follows – from 9am till 5.30pm lines/ports will be routed to In Office Hours destinations and from 5.31pm till 8.59am, the lines/ports will be routed to the Out of Office hour destinations.

Mobile Transfer Agent Service
This service allows users to manage forwarded calls to their mobile phones (GSM).

For the Mobile Transfer Service to work, the option "Ring my extension and my mobile at the same time" must be enabled in the extension’s forwarding rule. Other external numbers cannot be used to activate the mobile transfer agent. This service provides the ability to make transfers using DTMF inputs so you can effectively make transfers from your mobile phone. Available only in Pro or Enterprise Edition.

Feature 1: HOLD

While in a call press *80 - This will put the current call on hold.

Feature 2: UN-HOLD

Press *81 - this will un-hold a current held call.

Feature 3: Blind Transfer to an Extension

Press *82# number/extension # - This will make a blind transfer of your current call to the extension or number that you want to.

Example: Transfer a call to extension 105 or an external number 099219095 where 0 is the outbound rule for the call to go out via PSTN.

Dial *82#105# OR *82#099219095# - This will make a blind transfer (*82) of your current PBX call to the dialed extension (105) or number (099219095). Your mobile will disconnect from the call and the PBX call will be connected to 105 or 099219095 respectively.

Feature 4: Attended Transfer

Answer the incoming call from the PBX
Press *83# number/extension # to put the current call on hold and call that number
Press *84 to complete transfer once the recipient answers
Example: When you are in a call press *83#105# - This will put the current call (from the PBX) on hold and make a new call to 105. When 105 answers announce the call to the recipient and dial *84. The 3CX PBX will join the previous held call with 105 and disconnect you from both calls.

Feature 5: Conference

You can now create 3 way conferences using the Mobile Agent Service from your mobile phone. To create a 3 way conference follow the next steps:

Answer the incoming call from the PBX
Press *83#number/extension# to put the current call on hold and call that number
Once the number/extension is picked up dial *85. This will un-hold your previous held call, keep your mobile connected, and transfer the previously held call to the number that you dialed in step 2 thus creating a conference with 3 people
Example: accept incoming call and make a 3 way conference with number 105.

Answer incoming PBX call. Dial on your phone’s keypad *83#105# (this will put the previous call on hold and dial 105)

When 105 answers dial *85 to create a conference between 105, you, and the call you answered at step 1.

Block Outbound Caller ID
You can hide the outbound caller ID on a specific call. To do this just prepend the dialing number with *5. Example: if I want to call the number004412345678 but I want to do this anonymously, dial *5004412345678 or Outbound rule + *5 + Number. Your outbound caller id will be hidden.

Hotel - Maid Codes
Maid codes are specific codes which can tell PMS systems the status of a guest room in a hotel. This depends on PMS integration to be pre-configured. The codes vary for Mitel and Fidelio however the concept of how to control the status of the room is the same. For maid codes to work you must pre-configure an IVR to be a Wake-up Call IVR Service. (Go to Digital receptionist, edit an IVR and scroll all the way to the bottom where you will find an option named Wake-up Call IVR Service.)

 

Let's assume that a maid is in a hotel room and wants to set the room to Dirty/Vacant. From the phone in the room (room = extension number) dial *68 to set the room’s state. Connected PMS systems, such as Fidelio, will be informed to set the room status accordingly. An example for Fidelio, the standard code for Dirty/Vacant is “2”, would be *682. PMS connected users will see that this room is now Dirty/Vacant. The default room status codes supported by MITEL and Fidelio are:

MITEL Protocol:

Maid Present
Clean
Not Clean
Out of Service
To be inspected
Occupied/Clean
Occupied/Not Clean
Vacant/Clean
Vacant/Not Clean
Micros Fidelio:

Dirty/Vacant
Dirty/Occupied
Clean/Vacant
Clean/Occupied
Inspected/Vacant
Inspected/Occupied

3CX Phone System Parameters Table

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The Custom Parameters contains a list of advanced settings for 3CX. The default value of these parameters generally applies to most installations, and should not be altered unless otherwise advised to do so by 3CX Support.

The Custom Parameters can be found in the 3CX Management Console > Settings > Parameters > Custom Parameters tab. Below is a list of all parameters:

 

 

Name Description Value
ACPRMSET The name of currently active prompts set 8210986B-9412-497f-AD77-3A554F4A9BDB
ALERT_INFO_URL URL used for alert info so the phone can download the required ring tones from this url http://www.notused.invalidtld
ALLOW_CALL_TAKEOVER Allow takeover of established calls using Re-invites with replaces. Available options 0 off and 1 on 1
ALLOWFWDTOEXTFROMGRP Allows members of ring groups and queue to forward calls to external numbers. Requires that the gateway must be able to accept a call ONLY when the callee has answered – Early media OFF. Default enabled 1
ALLOWSOURCEASOUTBOUND Used only for remote extensions. If 1 (on) PBX saves the source IP:port of last successful OK to REGISTER message and constructs target requests to that IP:port, except those that have FQDN as target. If set to 0, ACK will be sent to IP:port specified in Contact header of 200/INV 1
ALLOWSOURCEASOUTBOUNDVP Used only for VOIPProviders. Default 0 If 1 (on) PBX saves the source IP:port of last successful OK to REGISTER message and constructs target requests to that IP:port, except those that have FQDN as target. If provider needs this to 1 to work, then this provider is unreliable 0
ALLOWUSEBUSYOPTFORGROUP =“1” will enable use of phone ‘Busy’ status instead of ‘PBX status’ for members of ring groups 0
ANALYSEVOIPPACKETS If set to 1, PBX will perform various consistency checks on VoIP Packets, and report results in server activity log. Useful when troubleshooting remote extensions 0
APNS_HOST FQDN of APPLE APNS Push Service. For development only use gateway.sandbox.push.apple.com gateway.push.apple.com
APNS_PORT Port of APPLE APNS Push Service 2195
APPENDCIDQUEUE enable 1
APPPATH Application Installed Path C:\Program Files\3CX PhoneSystem\
ATTACH_3CXCONFIG_WELCOMEEMAIL Parameter to attach 3cxconfig provisioning file to 3CX welcome email used to automatically configure 3CXPhone. Available options 0 off and 1 on 1
AUDIO_PROVIDER_ENABLE_AUTOGAIN 3CX Audio Service provider will dynamically adjust the volume of the stream in quiet parts to the level specified by parameter AUDIO_PROVIDER_MAX_VOLUME. Default value is 0 Disabled. Set to 1 to enable this option 0
AUDIO_PROVIDER_MAX_VOLUME Normalizes volume of the audio stream produced from Line in or Audio Playlist in case the audio files used do not have equal volume. Allows values are numeric and specify a percentage. Default is 0 – disabled. Example set to 25 to get -12dB 0
BILLFREQUENCY This parameter tells the billing module to round up to the next unit specified. Default behavior is blank. If it is set to 5 billing will round up to the next 5 units  
BILLINGCODE BillingCode dial code **
BILLINGROUNDUP This controls whether the rounding up for billing will be made in minutes or in seconds. Default behavior is blank in seconds. 1 = minutes  
BLACKLISTCHANGED The dummy parameter used for black list change event  
BLOCKREMOTETUNNEL Ignores failed authentication requests originating from tunnelled connections 0
BMACCEPTTOUT Boomerang acceptor timeout in seconds 5
BMCALLTOUT Boomerang caller timeout in seconds 120
BUILD Build version 670
BUSYPROMPT Replaces busy tone with Busy prompt. This will answer the call. 0 = busy signal on busy. 1 = Busy prompt played out when busy 0
CALL_TRANSFER_TRACKING_TIME Time interval in seconds in which Transferer will track status of blind transfer. If transfer will not be completed during this time, the transferer will be removed from the call and the result will depend only on the destination 60
CALLASSISTANTVERSION Current Version of the Call assistant Server 36097
CALLBYNAME_MAXLEN Maximal length of number for call by name -1
CALLBYNAME_MINLEN Minimal length of number for call by name 3
CALLHISTITERIVL callhistory Interval in seconds for processing iteration 60
CALLHISTTEMPLPERCALL Path to CallTemplate.xml for Per Call file output C:\ProgramData\3CX\Data\CDRTemplates\CDRTemplate-PerCall.xml
CALLHISTTEMPLSINGLE Path to CallTemplate.xml for single file output C:\ProgramData\3CX\Data\CDRTemplates\CDRTemplate-Single.xml
CALLHISTTEMPLSOCK Path to CallTemplate.xml for Socket text output C:\ProgramData\3CX\Data\CDRTemplates\CDRTemplate-Socket.xml
CALLHISTTEMPLSOCKLISTEN Path to CallTemplate.xml for listening Socket text output C:\ProgramData\3CX\Data\CDRTemplates\CDRTemplate-SocketListen.xml
CALLHISTTODB Flag to write callhistory to Database or not TRUE
CALLHISTWEEKLY Flag true=1 file per Week or false=per day TRUE
CALLRECORDSROOT Define the root path of call recordings. C:\ProgramData\3CX\Data\Recordings
CBTEST Call Back Test Extension, an internal 3CX Service Extension used for the 3CX Firewall Client Checker *888
CDM_EX_CONVERT Convert + to any value for contacts received from MS Exchange =+
CDM_EX_GAL Option to sync the first 100 Global Address Book Users from MS Exchange FALSE
CDM_EX_MAIL User Accounts to sync from MS Exchange  
CDM_EX_PASS Impersonated User Password  
CDM_EX_PUB Public Folders to sync from MS Exchange  
CDM_EX_TIME Time when syncronization will occur from MS Exchange. Default 4AM 400
CDM_EX_URL MS Exchange OWA URL  
CDM_EX_USER Impersonated User Account  
CDM_EX_VERSION Version of MS Exchange Server  
CDRENABLE Enabling of CDR records output 0 = disabled and 1 = enabled. Default=0 0
CHECK_CALL_DIVERSION_LOOP Algorithm to check for infinite system routing loops. If set to 1 Internal call history will be used for loop detection. If set to 0 loop detection works for forward all type rules 1
CONFERENCEEXTPIN Default Password Pin number for authentication for new created conferences Default value 0000 0
CONFPINLENGTH Controls the length of digits to generate the access PIN code to join a scheduled conference. Default value is 4 4
CONFPLACE_MOH_SOURCE Music file to be played to conference participants waiting in a conference C:\ProgramData\3CX\Data\Ivr\Prompts\onhold.wav
CONFPLACE_RTPFIRST RTP range of ports allocated for conference room 40310
CONFPLACE_RTPLAST RTP range of ports allocated for conference room 40438
CONFPLACE_SIPPORT specifies SIP port for Conference Room component. 40300
COUNTRYCODE Country index for e164 Number processing 357
COUNTRYINDEXNAME Country name for the country index Cyprus
CPSIPINTERFACE Binding address of NIC for conferenceplace component 127.0.0.1
CUSTOM1NAME Allows ability to change the mapping from Profile Available 2 to something else Available 2
CUSTOM2NAME Allows ability to change the mapping from Profile Out Of Office 2 to something else Out of Office 2
DEFAULTAUDIOFILE Default Music on hold file for Audio Feeder C:\ProgramData\3CX\Data\Ivr\Prompts\onhold.wav
DEFAULTHOLIDAYPROMPT Contains the path to the default holiday prompt used by the holiday service digital receptionist to play in case the specific holiday does not have a holiday prompt  
DEFAULTLOCALIP The default local IP of the PBX 192.168.9.57
DEVELOPMENT_DNTABLE_EXPOSE This will enable a DN property table in the advanced settings used for developers only. Available options Default 0 disabled 1 enabled 0
DIALCODECONFGATEWAY Dial code to access conference gateway service 700*
DIALCODEOUTOFFICE Deactivate Office Hours – Calendar settings and time used as set in the global Office hours in Settings/General.*641 IN Office *642 OUT OF Office irrespective of current time  
DIALCODEPROFILE Change profile using dial code Default *3. Options *30 Available *31 Away *32 OutOfOffice *33 Custom1 *34 Custom2 *3
DISABLE_OUTBOUND_IN_OOH Disables Outbound Calls in VP, Bridges, Gateways when Out of office hours is triggered. If 0 feature is disabled. If set to 1 Feature is enabled. Default 0 0
DISABLE_SYSCALL_AUTH This option controls authentication of INVITE requests from IVR (makecall), Queue (“polling” call) and Paging group (intercom call to member of paging group). 0 (default)=requires authentication. 1=does not require authentication 0
DISALLOWREFERTOBUSYSP Available options 0 and 1. 0=PBX will allow transfer request to a busy parking place. 1=PBX will reject transfer request to a busy parking place 0
DUMMYGLOBALOFFHOURSEV The dummy parameter used for office hours change event  
ECHOTEST Echo Test Extension used to test echo, delay and round trip timing *777
ENABLECONFERENCEPIN Enables the Conference Server Password Pin for authentication for new created conferences Default Off 0/On 1 0
ENABLEEARLYMEDIA Switch off Early Media Support on 1 or off 0 1
ENL Extension number length 3
EXPIRATIONGRACEPERIOD Expiry Grace Period 0
EXT_REFMT_INTL_PREFIX This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for INTERNATIONAL formatted calls  
EXT_REFMT_LOCAL_PREFIX This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for LOCAL formatted calls  
EXT_REFMT_NATL_PREFIX This prefix (outbound rule) will be added to dialled outgoing numbers from extensions for NATIONAL formatted calls  
FAXDIRECTSDP modifies behavior of SDP negotiation when the FAX extension is the destination 1
FAXOVEREMAILGATEWAY The default 3cx fax extension. The fax over mail extension 888
FIRST_STARTED internal parameter 6.35401E+17
FIRSTEXTPORT First port to use for external calls 9000
FIRSTLOCALPORT First port to use for internal calls 7000
FORCEREAUTH forces PBX to re-request authentication if it has nonce that has already been used. If value = 1, PBX will force the check. If set to 0 – PBX will not check reuse of nonce 1
GCM_AUTH GOOGLE GCM Google cloud messaging Push Service authentication key. Should be in the format key=followed by key from google account  
GCM_HOST FQDN of GOOGLE GCM Google cloud messaging Push Service https://android.googleapis.com/gcm/send
GCM_SENDERID GOOGLE GCM Google cloud messaging Push Service sender ID  
HIDECALLERID Block Outbound Caller ID *5
HIDECALLERIDINFO This information will be send in the INVITE instead of the Outbound Caller ID Information, Used in conjunction with HIDECALLERID parameter anonymous
INTERCOM Intercom dial code Default disabled  
INTERNATIONALPREFIX International Prefix 0
IVR_MOH_SOURCE Controlled by parameter IVR_USEBK_MUSIC. This specifies the source of the music to play in the background for IVR calls example name of file. Default blank  
IVR_RTPFIRST RTP range of ports allocated for IVR. Default 128 sim calls. 2 ports per call 40610
IVR_RTPLAST RTP range of ports allocated for IVR. Default 128 sim calls. 2 ports per call 40866
IVR_SIPPORT specifies SIP port for IVR component. 40600
IVR_USEBK_MUSIC Enable or disable background music for all IVR specified by parameter IVR_MOH_SOURCE. Available options 0=off 1=on 0
IVRDIALINGTONE Path to audio dilaing tone file that will be played during the first seconds of delay when call to IVR is made C:\ProgramData\3CX\Data\Ivr\Prompts\USProgresstone.wav
IVRDONOTCALL Comma separated list of destination numbers including ranges that will not be allowed for dialing from all IVR in 3CX. Default empty  
IVRNOCHECKFORDN If set to 0 IVR will block calls to extensions that do not exist in the system. If set to 1, IVR will allow calls to extensions that do not exist and they will be processed through the outbound rules 0
IVRPROMPTPATH Path for the ivr prompt file C:\ProgramData\3CX\Data\Ivr\Prompts
IVRSIPINTERFACE Binding address of NIC for IVR component 127.0.0.1
KEEPALIVE_TIME_UDP Controls interval in seconds that keepalives will be sent. Default is 30 seconds 30
LANGUAGE Installation language code en
LASTEXTPORT Last port to use for external calls 9049
LASTLOCALPORT Last port to use for internal calls 7499
LOCALSUBNETS Default Local Subnets reserved for Private Networks by the IETF and IANA RFC1918 10.0.0.0/8,169.254.0.0/16,172.16.0.0/12,192.168.0.0/16
LOGGEDINQUEUE LoggedInQueue dial code *62
LOGGEDOUTQUEUE LoggedOutQueue dial code *63
MAILBODYMISSEDCALLS Email Body for missed calls notification email You have a new missed call. From: %from% at: %start_time%. Ringing time: %ring_time% to extension:%to%
MAILSUBJMISSEDCALLS Email Subject used in Missed calls notification email New missed call from: %from% at: %start_time%
MAINTENANCE_CHECK_INTERVAL Interval in minutes to check for logs that can be deleted or moved to backup folder. Default 30 minutes 30
MAINTENANCE_CMBINLOG_MAXFILES Minimum number of 3CX Phone System Binary log files to keep in the log folder 2
MAINTENANCESCRIPTONSTARTUP Forces configuration service to run maintenance script on startup 1
MAXCALLDURATION This is the maximum timeout for connected calls Default is 3 hours. Calls will drop after this time 10800
MAXNOANSWERTIMEOUT This is the maximum timeout for ringing calls used for backup purposes – Default is 5 minutes 300
MAXNOOFDIDS Maximum Number of DID/DDI that can be created globally in the system 4000
MINVOICEMAILDURATION Voicemails that are less or equal to the value specified are not saved. Default value is 0 – disabled. Value is in milliseconds 0
MISSEDCALLSDATETIMEFORMAT Allows to change the format of the date.Example: yyyy/MM/dd HH:mm:ss For more informamtion check here: http://msdn.microsoft.com/en-us/library/8kb3ddd4.aspx If empty or incorrect, the current computer culture settings will be used.  
MISSEDCALLSRINGTIMEFORMAT Allows to change the format of the time for the duration of the missed call.Example: hh\:mm\:ss For more information see here http://msdn.microsoft.com/en-us/library/ee372287.aspx. If empty or incorrect the current computer time settings format will be used  
MLFROMADDR Senders e-mail address for notifications این آدرس ایمیل توسط spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
MLSRVADDR SMTP server address smtp.gmail.com
MLSRVPASSWORD Password for accessing SMTP server password
MLSRVPORT SMTP server port  
MLSRVTLS Enables SSL/TLS to send email notifications. Available values are 0 OFF and 1 ON. Default value is 0. 1
MLSRVUSER Username for accessing SMTP server این آدرس ایمیل توسط spambots حفاظت می شود. برای دیدن شما نیاز به جاوا اسکریپت دارید
MS_ALLOWVIDEOANDIMAGEPROXY Available options 0 off and 1 on to enable inbuilt video proxy. Re-invites must always be enabled for this to work 1
MSADDRFC2833FORINBANDDTMF Allows Media server to distribute recognized in-band DTMF as RFC2833 to other participants of the call 1
MSE_ADDRESS Address of the MS Exchange IVR  
MSE_USEIVR Using MS Exchange IVR 0
MSEXCH_SPECIALMENU Value is a string. You can enter the value MNU for compatability with Exchange 2007 or voicemail special number example 999. Default value is Blank which defaults to 999  
MSRTPTOS This parameter instructs the Media Server to mark all the packets it generates with a value between 0 and 63 in the DSCP field in the IP Header. Default Value: 0 – Valid Values: 0-63 Please note that Windows XP and Server 2003 disables IP Header tagging by default. You will need to set a DWORD registry value called DWORD registry value called DisableUserTOSSetting in the following location “HKEY_LOCAL_MACHINE > SYSTEM > CurrentControlSet > Services > Tcpip > Parameters” and give it value “0”. You will need to restart the machine for this setting to kick in. 0
MUSICONHOLDBEHAVIOUR Controls the behaviour of Music on Hold Default 0 OFF 0
MUSICONHOLDFILE Music file for music on hold C:\ProgramData\3CX\Data\Ivr\Prompts\onhold.wav
MYPHONE_SHAREDPARKING_DIALOG Enables the shared parking dialog in 3CXMyPhone. Available options 0 – disabled will not show transfer dialog and park to the first available shared parking place and 1 – Enabled. Display dialog and user can select a shared parking place 0
MYPHONECLIENTLOGGINGLEVEL This will control logging of 3CXMyPhone clients. Available options 0=disabled, 1=Low, 2=Medium, 3=Verbose 0
MYPHONESIPAUTH Hide or show SIP Authentication Tab in myPhone. 0 hide 1 show 0
MYPHONEVERSION 3CXPhone Version 12.0.36097.670
NEWAXFEREMULATION This enables new attended transfer functionality notifying the pbx when a transfer was successful for those phones that do not support replaces. Available options are 1=ON and 0=OFF 1
NONCEEXPIRATION defines nonce expiration time (seconds). By default it is 20 sec. After this time has passed since nonce issue — PBX will treat that nonce as expired and will force re-authentication (will issue 407 response with new nonce freshly generated. 20
NOTALLOWED_COUNTRYCODES Specifies the country codes which are not allowed for outbound calls 001, 001, 001264, 001268, 001242, 001246, 001441, 001345, 00506, 0053, 001767, 001809, 001829, 001849, 00503, 00299, 001473, 001671, 00502, 00509, 00504, 001876, 0052, 001664, 00505, 001670, 00507, 001787, 001939, 001869, 001758, 001784, 001868, 001649, 001284, 001340, 0054, 00297, 00501, 00591, 0055, 0056, 0057, 00593, 00500, 00594, 00590, 00592, 00596, 00595, 0051, 00508, 00597, 00598, 0058, 005993, 005994, 005995, 005997, 005998, 005999, 00355, 00376, 00374, 0043, 00375, 0032, 00387, 00359, 00385, 00420, 0045, 00372, 00298, 00358, 0033, 0049, 00350, 0030, 00441481, 0036, 00354, 00353, 00441624, 0039, 00441534, 00371, 00423, 00370, 00352, 00389, 00356, 00373, 00377, 00382, 0031, 0047, 0048, 00351, 0040, 007, 00378, 00381, 00421, 00386, 0034, 004779, 0046, 0041, 0090, 0090392, 00380, 0044, 003906698, 0093, 00994, 00973, 00880, 00975, 00673, 00855, 0086, 00670, 00995, 00852, 0091, 0062, 0098, 00964, 00972, 0081, 00962, 00965, 00996, 00856, 00961, 00853, 0060, 00960, 00976, 0095, 00977, 00850, 00968, 0092, 00970, 0063, 00974, 00966, 0065, 0082, 0094, 00963, 00886, 00992, 0066, 00993, 00971, 00998, 0084, 00967, 00213, 00244, 00247, 00229, 00267, 00246, 00226, 00257, 00237, 00238, 00236, 00235, 00269, 00242, 00243, 00225, 00253, 0020, 00240, 00291, 00251, 00241, 00220, 00233, 00224, 00245, 00254, 00266, 00231, 00218, 00261, 00265, 00223, 00222, 00230, 00262269, 00269639, 00212, 00258, 00264, 00227, 00234, 00262, 00250, 00290, 00239, 00221, 00248, 00232, 00252, 0027, 00211, 00249, 00268, 00255, 00228, 002908, 00216, 00256, 00260, 0025524, 00263, 001684, 0061, 00614, 006189162, 00682, 00679, 00689, 00686, 00692, 00691, 00674, 00687, 0064, 00683, 00672, 00680, 00675, 00872, 00685, 00677, 00690, 00676, 00688, 00699, 00678, 00681
OPERATOR Operator extension number 100
PARK Park dial code *0
PARK_MOH_SOURCE Overrides default path/name of the Music on hold file to be played for all Parked Calls. Default is blank C:\ProgramData\3CX\Data\Ivr\Prompts\onhold.wav
PARKORB_RTPFIRST RTP range of ports allocated for Parking Orbit 40010
PARKORB_RTPLAST RTP range of ports allocated for Parking Orbit 40138
PARKORB_SIPPORT specifies SIP port for ParkingOrbit omponent. 40000
PBXERRORCODES Defines the email notification error codes  
PBXPUBLICIP The public Ip/domain of the PBX 3.3.3.11
PEC Prefix for external call 0
PERS_PHBK_ONLY System wide option. Currently available for Polycom only. Available values 0 1. 1 removes company phonebook leaving personal phonebook only. 0
PHONEBOOK_LAST_FIRST Defines caller name replacement format. Available values 0=FirstName LastName 1=LastName FirstName Default 1 1
PHONEBOOK_MIN_MATCH Available values 0 disabled,-1 exact match only, N>0 minimum match length required for best-match strategy, numbers are compared from tail to head 0
PHONEDEVICESVERSION Version number of Phone Devices Auto Detection 36097
PICKUP Pickup dial code *20*
POSIPINTERFACE Binding address of NIC for Parking orbit component 127.0.0.1
PREPENDNAMETOCID name of queue / inbound rule will be pre-pended rather than appended  
PRMUPDTM Time of the last changing within the prompts settings  
QM_RTPFIRST RTP range of ports allocated for Queue Manager 32000
QM_RTPLAST RTP range of ports allocated for Queue Manager 32399
RANDOMMUSICPERCALL Randomizes Music on Hold per call made Default value 0 OFF ie MOH is randomized Per Day. When 1 option is on and MOH will be randomized per call 0
REGISTER_REQUEST_MAX_TIMER Controls interval of registration period when 3CX is answered with 503 Service Unavailable. Default Re-registration is 120 seconds (2 minute) Value is in seconds 120
REMOVECOUNTRYCODE Remove if same Country 1
REP_CMDTIMEOUT This is the max time allocated to the reporter to query the database for generation of reports. On large systems should be increased if exceptions are detected. Default time in seconds 900
RESPONSE_ERROR_CODES These are all the error codes that will be reported with Event ID 12294 when received from VoIP Prov Bridge PSTN Trunk. Comma separated list of sip error codes 400, 401, 403, 404, 405, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000, 000
SEC_BLACKLIST_TIME Amount in seconds to keep the abusing IP address in blacklist. Default value in seconds 86400
SEC_IGNORE_USER_AGENT List of user agents in comma separated form that are by default used in security and hacking tools. 3CX will immediately ignore any responses and act dead friendly-scanner,sipsak,smap,Elite 1.0 Brcm Callctrl,sipcli
SEC_MAX_CHALLENGES When IP Challenges 3CX and does not answer to 407 with authentication, these requests are counted per IP and when value is reached, IP is blacklisted for Blacklist time specified 1000
SEC_MAX_FAILED_AUTH Authentication protection. If attacker,device sends failed register/proxy auth required, entity is blacklisted. SEC_BLACKLIST_TIME parameter will be loaded. Default value 25 25
SEC_RATE_BARRIER_1 Protection Layer 1 Triggered when Server receives X pkts/sec. Further requests will be rejected and will respond with 503 to many requests-resend after value SEC_RETRY_AFTER parameter 2000
SEC_RATE_BARRIER_2 Protection Layer 2 triggers when layer 1 is abused. Default value in Packets per second. Abusing IP will be immediately blacklisted for value specified in SEC_BLACKLIST_TIME parameter 4000
SEC_RATE_TIME_ALLOW Allowed free time – Abuse counting starts but no checks are made in this period – After this period expires, checks are enforced. Default value in ms 200
SEC_RETRY_AFTER Value specified in 503 Too many requests triggered for Barrier 1 responses. Default value in seconds 5
SECURESIPPATH Secure SIP Certificate Path C:\ProgramData\3CX\Bin\Cert
SETAVAILABLE SetAvailable dial code *60
SETAWAY SetAway dial code *61
SHOWEXTWEAKPASS Enabled by default – Highlights in red extensions that have weak passwords and voicemail pin numbers 1
SHOWMUI Parameter to enable 3CX Webmeeting user interface 1
SINGLE_SIP_IP Available options 0 off default behavior, 1 on which means that pbx will use ip specified in parameter SIPINTERFACEIP in all SIP contructs instead of calling windows function GetBestRoute 0
SIP_FORKED_ID_BUSY Available options 0 and 1. If set to 0, when one of the sip forked entities sends a busy signal, all invites to all participants are canceled at once. If set to 1, each participant must send a busy respectively. Other SiP Forked devices remain on target 1
SIPDOMAIN SIP Domain Name 192.168.9.57
SIPINTERFACEIP IP Address that phone system server will use in all SIP negotiations. If IP is specified (must be present on stack) and parameter SINGLE_SIP_IP is set to 1, call manager will use this IP in all SIP constructs overriding windows function GetBestRoute. Default blank  
SIPPINGPERIOD Time intervall to poll endpoints with SIP INFO messages, in seconds. 0 – disabled 0
SIPPORT Sip Port 5060
SLA_ENABLED Enable or disable SLA feature – Available values 0 off or 1 on Default Value Off 0
SLA_ONSUBSCRIBE Defines how many SLA slots are available for each shared appearance when SLA is activated by device subscription. 0 means none. Positive N means N slots are allocated on the first subscription 0
SMDRHOSTPORT SMDR Listening host and port, empty host = disabled(default), use 0.0.0.0:1752  
SNOM_TBOOK Parameter to control SNOM phonebook. Available options are true and false. When true, the phonebook directory is erased and re-provisioned from scratch. Default is true TRUE
SP_VERSDESC The installed service pack version description SP.
SPP_DROP_PICKUP_CALLS Shared parking place will forcedly terminate simultaneous attempts to pickup a call from multiple Cisco SPA Phones simultaneously. Available options are 1=ON 0=OFF. Default Off 0
SPUPDATE_EMAIL_NOTIFICATION Sends an email notification to the administrator when there is a new service pack update 0
STUNDISABLED Flag used to inform PBX about the Stun disabled status 0
STUNSRV Default STUN server address/port stun.3cx.com
STUNSRV2 Default second STUN server address/port stun2.3cx.com
STUNSRV2PORT Default second STUN server port 3478
STUNSRV3 Default third STUN server address/port stun3.3cx.com
STUNSRV3PORT Default third STUN server port 3478
STUNSRVPORT Default STUN server port 3478
STUNTCHECK Recheck external IP every, sec 1200
STUNTOUT Timeout time for STUN response, ms 3000
SYSTEMLOGGINGLEVEL The current logging level 2
TIME_DST_END_DAY Day when DST settings end 25
TIME_DST_END_HOUR Hour when DST settings end 3
TIME_DST_END_MONTH Month when DST settings end 10
TIME_DST_SAVING_MINUTES Number of minutes to add while DST is active 60
TIME_DST_START_DAY Day when DST settings start 28
TIME_DST_START_HOUR Hour when DST settings start 2
TIME_DST_START_MONTH Month when DST settings start 3
TIME_NTP_SERVER NTP server pool.ntp.org
TIME_TIMEZONE_AASTRA Timezone in Aastra-compatible format US-Eastern
TIME_TIMEZONE_CISCO79X0 Timezone for Cisco 7940 7960 compatible format EST
TIME_TIMEZONE_CISCO79X1 Timezone for Cisco 7941 7961 compatible format Eastern Standard/Daylight Time
TIME_TIMEZONE_FANVIL Timezone in Fanvil compatible format 14
TIME_TIMEZONE_GRANDSTREAM Timezone in Grandstream-compatible format 420
TIME_TIMEZONE_GRANDSTREAMEXEC Timezone in Grandstream-compatible format for new Grandstream Executive phone models EST5EDT
TIME_TIMEZONE_LINKSYS Timezone in Linksys-compatible format GMT-05:00
TIME_TIMEZONE_SNOM Timezone in Snom-compatible format USA-5
TIME_TIMEZONE_YEALINK Set timezone for Yealink – Available options are -13 to 12 Default -5 US Eastern Standard Time -5
TIME_UTC_OFFSET_MINUTES Difference from UTC in minutes 60
TIME_UTC_OFFSET_SECONDS Difference from UTC in seconds- Used for Polycom phones -18000
UNPARK Unpark dial code *1
USERAGENTSTRING User Agent that 3CX will send can be customized. This will hide the identity of the VoIP PBX so attackers cannot know what is behind port 5060 Needs full restart of services  
VERSION Version 12.0.36097.670
VMAIL VMail dial code *4
VMDIALOUTENABLED Enable dialout calls from voicemail menu. By default it is 0. Set to 1 to enable dialout calls. 0
VMN_RETRY_MINUTES Time interval in minutes to resend email notifications 15
VMN_STOP_RETRY_HOURS Time interval in hours to stop sending a failed email notification Default 3 days 72
VMN_TEMPLATE Path where Voice Mail Email Notification template is located. This will be used to send voicemail email notifications upon receipt of a voicemail. Template name is VMNTemplate.xml C:\ProgramData\3CX\Data\Http\Templates\VMNTemplate.xml
VMPINREQUIRED Determines whether VM will ask for pin. By default it is 1. Set to 0 to access the VM menu directly(e.g. Hotel extensions). 1
WCS_PASSWORD Administrator Password used to access 3CX Web Conferencing Server  
WCS_USER Administrator User Name used to access 3CX Web Conferencing Server  
WEBSERVER Webserver used in 3CX Phone System ABYSS
WEBSERVERPASS The admin pass admin
WEBSERVERUSER The admin ID admin
WELCOME_EMAIL_EXT_AGENT Email body to be sent when a welcome email is sent to an external queue agent. Default is in English  

 

https://www.3cx.com/docs/parameters-table/

 

Common Mistakes Which Lead to Call Fraud

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

Although a rare occurrence, we have seen a few cases of call fraud where several misconfigurations led to the unnecessary exposure and compromisation of 3CX. In this blog post we are going to highlight some common mistakes we’ve noticed that lead to call fraud and what you can do to avoid them.

What is a Call Fraud?
Simply put, a call fraud occurs when untrusted parties place calls through your PBX, at your expense. Usually this happens overnight or when offices are closed, and calls are placed in bulk to various international destinations. And then you get a large bill at the end of the month which you need to pay….

Since the early ages of PBXs, free international calling has been the target of phone phreaks. You might think that it’s something of the past with communication costs having drastically decreased, however in the modern age of VoIP telephony a global threat of organized crime has occurred looking to make big profits in an industrial manner.

Usually, a hacker compromises IP PBX servers in order to establish calls to premium international numbers. His motivation is an indirect financial gain, as he will dial thousands of numbers in an automated manner from premium services under his control, in order to get paid commissions per call or per time spent on the line. This is also known as International Revenue Sharing Fraud (IRSF). Another common way to profit more directly is the simple resale of stolen credentials on the darknet for whoever wants a cheap route to dial out.

3CX Phone System has many inbuilt security features and default settings which prevent such abuses, however, administrators sometimes disable safeties without understanding the risks implied leading to the inevitable.

We will detail below the TOP 5 common mistakes to avoid.

Number 1: Weak Credentials
The first mistake is using weak credentials for your extensions.

When creating an extension in your phone system, default random credentials are generated at all levels, strong SIP Authentication ID and password for SIP, strong password for your web client, for your hard phone web interface, random voicemail PIN, etc. You should stick to those random values which ensure protection against password-guessing attacks, also called brute-force.

Since v15.5 it is impossible to edit and save an extension with credentials that are too short, however, you may have inherited such a setting from previous versions or backups.

We have also made sure to warn admins when they have weak credentials by signaling a warning flag next to extensions names. If you hover over the extension you will get more information on the issue:

Weak Extension Password Warning
By the way, please never set temporary credentials for testing, thinking that you will change them later when going in production as usually people tend to forget these things.

Number 2: Allowing Remote Access
The second most common mistake is to have the option “Disallow use outside LAN” unticked under your extensions when not needed.

This option prevents remote SIP registration on your extension and is ticked by default when creating an extension. You can still use a remote 3CX client or the 3CX Web Client under this condition without being affected, as the client uses the tunnel protocol to connect to the PBX. In effect, the option should be unticked only when using a remote STUN hard phone.

Do not allow use of extensions outside the LAN
Number 3: Too Many Countries Allowed
When first installing your PBX a screen prompted for the countries to allow in and outbound calls. This list can be later found in Security / Allowed country codes.

It should be restricted to the countries called commonly by the users. By default we restrict to the country of installation only.

A bad practice is of course to allow all countries thinking that it will be adjusted later, usually never.

Restrict the countries users are allowed to call
Note for US customers: the North American Numbering Plan (NANP) allows to dial 25 regions or countries from North America and the Caribbean, without an international dial code. The international dial code would be 011, or + in case of US. The anti-hacking feature will let such numbers go through as local numbers (as per ITU standards). You should therefore have strict outbound rules, with a list of NANP prefixes to block and route 1: Block calls, and ensure this rule is in first position.

Number 4: Lazy Outbound Rules
Another bad practice is to have “lazy” outbound rules, letting any number dialed from anyone in the system go through. A typical rule is one with no criteria other than the DEFAULT extension group.

You should have rules set as strictly as possible, like in firewalls, defining specific prefixes or number lengths, and which extensions or which extension groups will be allowed to dial out.

Number 5: Misconfigured E164 Settings
Under Settings / E164 Processing are standard settings ensuring the replacement of the “+” by your local international dial code. The reference is the country defined at the installation time. For example in most countries you will get “00” as the international dial code, for the US you will get “011”. Those are values as per ITU standards.

This setting is important as it is also used to determine the list of country codes blocked as per the “Allowed country codes” tab discussed above.
For instance, with “00” and Albania blocked, the feature will look for numbers dialed in the form of 00355xxx or +355xxx.
If you misconfigured the international dial code, this can result in a wrong “+” replacement but also in the safety being inoperant.

International Phone Number Configuration
Note that in most call fraud schemes observed, a cumulation of the previous 5 mistakes resulted in compromisation. One alone would usually not be enough for an attacker to be successful.

In v16 we introduced two major security features improving the security posture of 3CX even more. The first one allows you to restrict management console access based on the IP and is in Security / Security Settings / Console Restrictions. By default all IPs are allowed, if enabled it will let through only local IP subnets and specified public IPs. This option does not interfere with other web services such as provisioning, web client, etc.

Restrict your management console access based on IP Address

The second major improvement is the Automatic Global 3CX IP Blacklist, available in Security / Security Settings / Anti-Hacking. When this is enabled, your PBX will report any blacklist event including the attacker IP, to our centralized server. After evaluation, recurring attackers will be added and spread across all 3CX systems which have this feature enabled so that any malicious traffic will be dropped. To date this global blacklist has already got 1000+ common IPs and ranges which have been reported as scanning or frauding. We encourage you to enable this feature.

Automatic Global 3CX IP Blacklisting

If you have any security questions or if you wish to report a fraud for us to review and advise on what may have occurred and how this could have been prevented, feel free to open a Support ticket in our Helpdesk, under the “Security | Fraud” category. Those tickets are treated with High priority.

In case of an incident do not panic. It is important that you gather logs before taking any further actions so that forensics are archived. To do that, go to Support / Generate support information, which will generate a zip file and send you the link by mail. You can then attach this zip file to your ticket for review.

We look forward to receiving your comments on this topic!

 

https://www.3cx.com/blog/voip-howto/call-fraud/

 

List of Events Generated by 3CX Phone System

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

3CX Phone System will log certain events in the Windows Event Log and in the 3CX Server Event Log within the 3CX Phone System Management Console. Together with the 3CX Server Activity Log, this information can be used to troubleshoot and verify the performance of 3CX Phone System.

The following is a list of events that can be generated and logged by 3CX Phone System.

Event ID Event Logged Comments
102 One of the queues has initiated a callback
2013 The disk is at or near capacity (Windows Server OS Only)
4097 Service started Service has been started
4098 Service got signal to stop: %1 Service is about to stop
4099 Emergency number (%1) is dialed from %2 An emergency number was dialed
4100 Trunk %1 has changed status to %2 The status of a trunk has changed
4101 Extension %1 is %2%3 The registration status of an extension has changed
8193 Licence limit is reached, active calls: %1 A call request was received in excess of the Licence Key limit of simultaneous calls
12289 Device %1 had no available outgoing trunk(s) for Call(%2) An outbound call request over a trunk failed because over the maximum number of calls available through that trunk
12290 The IP %1 has been blacklisted for %2 sec. Reason: %3 An IP Address has been blacklisted
12291 SIP request (%1) from %2 was rejected. Reason: %3. Message: %4
12292 The IP %1 has been blacklisted for %2 sec. Reason: requests rate is too high! A request was rejected/blocked by the Anti-Hacking module – security breach detected
12293 Registration at %1 has failed. Destination (%2) is not reachable, DNS error resolving FQDN, or service is not available. DNS resolution failed or cannot reach remote VoIP Provider
12294 Call to %1 has failed. %2 replied: %3 Destination responded with an error code (eg. service unavailable, user not found, insufficient credit, account disabled) – contact trunk provider
12295 STUN server %1:%2 could not be reached%3 It could be that the STUN server you are using is down or a network problem is preventing STUN resolutions. This might cause problems if you use VoIP Providers. STUN resolution has failed, timed out, or no backup STUN servers could resolve
12296 Call from [%2] to [%1] has been rejected by the 3CX Country Blocking Feature [%3]. Reason: %2 contains Prefix %3. Calls to %3 are not allowed by system. Resolution: To allow this call access the 3CX Management Console, Settings, General, and enable the country or continent that matches this prefix.

https://www.3cx.com/blog/docs/list-of-events-generated-by-3cx-phone-system/

 

باز کردن پورت رنج ۱۰۶۰۰ الی ۱۰۹۹۸ در فایروال لینوکس دبیان

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 ​Please note that the firewall checker only checks 10600-10998 ports but the ports used for webrtc (webclient calls) are 10500-10998.
Additionally, if you have a voip provider configured on the PBX and/or remote STUN phones, you need to have the rest of the range opened (9000-10998).
Furthermore, please note that these ports need to be opened on the firewall as well if there is a firewall in front of the PBX.

 iptables -I INPUT -p udp --dport 10500:10998 -j ACCEPT

iptables-save > /etc/iptables/rules.v4

باز کردن پورت رنج ۱۰۶۰۰ الی ۱۰۹۹۸ در فایروال لینوکس دبیان - 5.0 out of 5 based on 1 vote

اتصال مرکز تلفن به EveryoneAPI

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

Introduction


3CX provides integration with EveryoneAPI. The configuration is done server-side, in an easy and straightforward way. This integration provides these benefits to users of 3CX and EveryoneAPI:

  • Contact Synchronization – Inbound calls from external numbers trigger a contact lookup in your CRM, and contact details are added to 3CX Contacts. This way, the caller name is automatically shown in your phone display when you receive the call.

  • Call Pop-ups – When using the 3CX Web Client, the customer record is brought up to you automatically when you receive an inbound call.

This guide takes you through the steps required to setup EveryoneAPI with 3CX server-side.

3CX CRM Server Side Configuration

  1. Create an account with EveryoneAPI https://www.everyoneapi.com/sign-up
  2. Login to EveryoneAPI https://www.everyoneapi.com/login
    EveryoneAPI Integration pbx
  3. Go to the “Dashboard > Account Details”, and take note of the Account SID and Authentication Token.
  4. Login to the 3CX management console > “Settings > CRM Integration > Server Side” tab.
  5. Download the EveryoneAPI provider template from the “Available CRM Integrations” section.
  6. Select the EveryoneAPI integration from the dropdown list.
    EveryoneAPI Integration ippbx
  7. Copy the Account SID and Auth Token in the respective fields.
  8. When you receive a call via the web client, a contact query is made to EveryoneAPI and the resolved name is displayed on the Web Client’s toaster.

کدک یا Codec چیست؟

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

کدک یا Codec چیست؟

کدک یا Codec مخفف Compressor Decompressor به معنی فشرده کننده و باز کننده است.

حتما کدک‌ بسیار محبوب mp3  را با آن آشنایی دارید و برای فشرده کردن فایل‌های صوتی از آن استفاده می‌شد.

کدک ه وظیفه تبدیل صدا از آنالوگ به دیجیتال و یا IP را دارند.

ادامه مطلب: کدک یا Codec چیست؟

تنظیمات مرکز تلفن ویپ 3cx و elastix برای اولین بار

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

تنظیمات مرکز تلفن ویپ 3cx و elastix برای اولین بار

First Time Configuration of 3CX and Elatix

  • New Install or Restore, Admin User and IP Type
  • Choose an FQDN
  • DNS type and Port selectio
  • Extension Length, Admin Email, Language 
ادامه مطلب: تنظیمات مرکز تلفن ویپ 3cx و elastix برای اولین بار

آموزش نصب و راه اندازی 3CX و elastix پس از نصب سیستم عامل لینوکس Debian

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آموزش نصب و راه اندازی 3CX و elastix پس از نصب سیستم عامل لینوکس Debian 

مقدمه 

این آموزش نصب و راه اندازی در زمانی مورد استفاده قرار میگیرد که شما سیستم عامل لینوکس خود را نصب کرده اید و تصمیم به نصب مرکز تلفن ویپ الستیکس یا 3cx دارید.

ادامه مطلب: آموزش نصب و راه اندازی 3CX و elastix پس از نصب سیستم عامل لینوکس Debian
آموزش نصب و راه اندازی 3CX و elastix پس از نصب سیستم عامل لینوکس Debian - 5.0 out of 5 based on 1 vote

برگه نمونه نصب و راه اندازی مرکز تلفن ویپ 3CX

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

برگه نمونه نصب و راه اندازی مرکز تلفن ویپ 3CX

 

Example Phone System Deployment Sheet
Preparation is key for deployment. Having the equipment and tools ready is key for looking professional and for getting the job done. It is helpful to have a small checklist or sheet to work from to ensure everything is prepared. The below is an overview of what you should look for before going ahead with the installation. You can use it as a sample sheet and depending on the installation you may need to add or remove from the below list.

Customer information
Basic customer information is a good starting off point. Key dates, carriers and other base-level information is helpful to keep things organized.

Company name and address
Carrier name
Install date
Server install date
Training date
Server IP address
Extension list
Especially for larger customers, a comprehensive list of extensions is important to deploying the right information. Generally, it is best to get the customer to fill this out because any errors will not be your fault.

Name
Ext
DID
Email
Vmail to email
Desk device
Home device
Softphone
Trunks
Document the different types of trunks you will configure:

Trunk name
Technology
Channels
DID 800
Outbound order
Ring groups
Ring groups are present in almost all deployments and having accurate information is needed to keep things running smoothly. This section keeps that in order.

Group name
Group #
Ring strategy
Ring time
Announcement
CID prefix
Failover destination
Members
Call queues
Not all systems require call queues but being prepared will save you time in deployment.

Queue name
Ring strategy
Static agents
Failover
Wait time

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مکان ذخیره فکس های دریافتی 3CX به صورت فایل PDF  در مسیر ذیل میباشد:

در سیستم عامل ویندوز:

​C:\ProgramData\3CX\Instance1\Data\Fax\Inbox

در سیستم عامل لینوکس:

​/var/lib/3cxpbx/Instance1/Data/Fax/Inbox

محل ذخیره فکس های دریافتی 3CX به صورت فایل PDF - 5.0 out of 5 based on 1 vote

دانلود 3cx

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دانلود مرکز تلفن نرم افزاری ویپ 3CX

3CX Phone System  یک نرم افزار مبتنی بر ویندوز و لینوکس است که با استاندارد SIP مبتنی بر VoIP کار می کند. این نرم افزار توسط بیش از 300,000 شرکت در سطح جهان استفاده شده است, 3CX یک شرکت مخابراتی برپایه فناوری و نوآوری شناخته شده است.

لینک دانلود نرم افزار سرور مرکز تلفن 3CX

Download 3CX server لینک دانلود نرم افزار سرور مرکز تلفن 3CX



3cx sbc download windows

دانلود نرم افزار 3CX SBC نسخه ویندوز

Download the 3CX SBC

 3CXSBC16.msi




call flow designer

لینک دانلود طراحی گردش کار و مکالمات تلفنی

3CX Call flow designer یا CFD

لینک دانلود طراحی گردش کار و مکالمات تلفنی 3CX Call flow designer یا CFD

Download the 3CX Call flow designer

دانلود 3cx - 5.0 out of 5 based on 3 votes

دانلود و بروزرسانی تلفن نرم افزاری 3cx

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ستاره فعالستاره فعالستاره فعالستاره فعالستاره فعال

دانلود و بروزرسانی تلفن نرم افزاری 3cx

جهت بهره مندی از تمامی امکانات تلفن نرم افزاری 3cx لطفا همیشه این برنامه را بروزرسانی نمایید.

تلفن نرم افزاری ویپ 3cx  برای سیستم عامل های ویندوز ، مک ، و تلفن های هوشمند و تبلتهای IOS و اندروید برای دانلود در دسترس میباشد.

لینک دانلود تلفن نرم افزاری 3cx 

3CX Clients Download Links

 

دانلود آخرین نسخه تلفن نرم افزاری 3cx  و بروز رسانی آن موجب افزایش سرعت صدای بهتر و امکانات بیشتر میگردد.

همچنین این تلفن نرم افزاری در نسخه V15.5 Update 6امکانات ذیل را دارا میباشد.

  • اضافه شدن کدک  OPUS  
  • سرعت اتصال سریع
  • پشتیبانی از یک داخلی برای چند دستگاه به صورت همزمان
  • تضمین دریافت اعلان های PUSH 
  • نمایش وضعیت سایر داخلیها به شما
  • جستجو در فهرست تماسهاها و دفترچه تلفن
  • QR Provisioning
  • Schedule Conference
  • Send eMail to scheduled conference participants
  • IPV6 support
  • Chat
  • unread message counter
  • call back from missed call notifications

نمایش ویديو معرفی تلفن نرم افزاری ویپ 3cx  نسخه اندروید:

نمایش ویديو معرفی تلفن نرم افزاری ویپ 3cx  نسخه اپل IOS:

دانلود و بروزرسانی تلفن نرم افزاری 3cx - 5.0 out of 5 based on 1 vote
کدک یا Opus چیست؟

اضافه نمودن کدکهای صوتی مانند GSM-FR , G729 , G.711 U-law , G.711 A-law , G722 , OPUS

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اضافه نمودن کدکهای صوتی مانند GSM-FR , G729 , G.711 U-law , G.711 A-law , G722 , OPUS

مطالعه بیشتر : کدک یا Codec چیست؟

جهت اضافه نمودن کدکهای صوتی مانند GSM-FR , G729 , G.711 U-law , G.711 A-law , G722 , OPUS به روش زیر عمل نمایید.

وارد تنضیمات داخلی مورد نظر شوید و در قسمت Phone Provisioning شوید.

در قسمت Codecs با زدن دکمه Add  کدکهای صوتی خود را به داخلی تلفنی خود اضافه نمایید.

شما میتوانید با بالا بردن کدک های صوتی Codec اولویت انتخاب کدک را انجام دهید.

opus codecs up or down to prioritize them 3CX

اضافه نمودن کدکهای صوتی مانند GSM-FR , G729 , G.711 U-law , G.711 A-law , G722 , OPUS - 5.0 out of 5 based on 1 vote

مکالمه صوتی و تصویری تلفنی تحت وب در 3cx و الستیکس

ستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعالستاره غیر فعال

امکان مکالمه صوتی و تصویری تحت وب در الستیکس ۵ و یا 3cx در نسخه ۱۵.۵ سرویس پک ۶ فراهم شده است.

لطفا جهت فعال سازی قابلیت ( مکالمه صوتی و تصویری تحت وب در الستیکس ۵ و یا 3cx ) وارد قسمت :

Settings > PBX > General > Enable WebRTC softphone in Web Client

ادامه مطلب: مکالمه صوتی و تصویری تلفنی تحت وب در 3cx و الستیکس

دریافت لایسنس رایگان مرکز تلفن 3CX و الستیکس ۵

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ستاره فعالستاره فعالستاره فعالستاره فعالستاره فعال

لایسنس رایگان مرکز تلفن ویپ 3CX یا همان الستیکس نسخه ۵ ،  نسخه استاندارد میباشد و امکان مکالمه تا ۱۶ تماس تلفنی همزمان را برای شما دارد.

محدودیتی در تعداد تلفن های داخلی و اتصال خطوط تلفن شهری نیست ، ولی فقط  ولی با استفاده از این لایسنس امکان ۱۶ تماس همزمان وجود دارد.

معمولا ۲۰٪ تلفن های داخلی در حال مکالمه همزمان میباشد. به عنوان مثال برای یک شرکت با ۶۴ تلفن داخلی لایسنس ۱۶ تماس همزمان کافی میباشد. 

شما میتوانید در قسمت Setting مرکز تلفن خود تنظیم نمایید در صورت افزایش بیش از حد نصاب لایسنس به شما توسط ایمیل اطلاع دهد.

لطفا توجه داشته باشید ۴۵ روز اول این لایسنس به صورت پروفشنال میباشد و امکانات نسخه پروفشنال پس از ۴۵ روز برداشته میشود و به استاندارد تبدیل میشود.

در صورت تمایل به داشتن امکانات نسخه پروفشنال مرکز تلفن مانند API و صف تماس پیشرفته یا Call senter میتوانید لایسنس را به لحظه از واحد فروش پارسه پلاس خرید نمایید.

جهت دریافت لایسنس مرکز تلفن ویپ 3CX آلمان به آدرس زیر مراجعه نمایید.

با توجه به اینکه وب سایت شرکت 3CX در سرورهای Google Cloud  است و سرویس ابری گوگل به سمت ایران بسته است ( به دلیل تحریمهای جاری گوگل) لطفا از فیل-تر-*شک-ن استفاده نمایید. 

 https://www.3cx.com/phone-system/download-phone-system?resellerId=241806

 


آموزش دریافت لایسنس ۱۶ تماس مرکز تلفن 3CX

فرمی به صورت زیر مشاهده مینمایید.

دریافت لایسنس رایگان مرکز تلفن 3CX و الستیکس ۵

با انتخاب گزینه Individual فرم را بصورت شخصی و با انتخاب Company  فرم را به صورت شرکتی تکمیل مینمایید.

نام خود را در فیلد Name  وارد نمایید.

نام خانوادگی خود را در فیلد Surname  وارد نمایید.

شماره تلفن خود را در فیلد Telephone  وارد نمایید.

کدپستی  خود را در فیلد Area or ZIP Code   وارد نمایید. با ورود ابتدای کد پستی خود میتوان از آدرسهای پیشنهادی گوگل استفاده نمود.

آدرس  ایمیل خود را در فیلد E-Mail  وارد نمایید. تایید کد لایسنس به این آدرس ایمیل ارسال خواهد شد.

 تعداد داخلی های خود را در فیلد Number of Extensions  وارد نمایید. این تعداد داخلی به نوع و نحوه عملکرد لایسنس شما تاثیر نخواهد داشت.

 هر دو تیک گزینه های قابل انتخاب زیر را بزنید و نهایتا بر روی دکمه Submit & Download  کلیک کنید.

لطفا ایمیل خود را چک نمایید. اکنون ایمیلی از طرف 3CX با موضوع Verify your Email & Get Started  بدست شما رسیده است.

ممکن است ایمیل ارسال شده به فولدر Spam یا Junk-mail  شما رفته باشد.

Verify your Email 3cx min

لطفا بر روی دکمه Verify your Email جهت تایید آدرس ایمیل خود کلیک نمایید.

کد لایسنس در صفحه ای که برای شما باز میشود به شکل زیر آمده است. کد لایسنس را کپی نموده و آن را در فرم  ارسال نمایید.

Use Your licence key

دریافت لایسنس رایگان مرکز تلفن 3CX و الستیکس ۵ - 5.0 out of 5 based on 3 votes

بروز رسانی موکز تلفن ویپ 3cx و الستیکس ۵ به صورت Command Line

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بروز رسانی موکز تلفن ویپ 3cx و الستیکس ۵ به صورت Command Line

جهت بروز رسانی مرکز تلفن ویپ الستیکس و 3cx  از طریق خط فرمان یا Command Line  در سیستم عامل لینوکس دبیان debian  از دستورات زیر استفاده نمایید.

 

1. wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -

2. apt-get update

3. apt-get upgrade 3cxpbx


The first command installs the key, the second command updates the repository information and lastly, the third command will upgrade your pbx to the latest release.

{rscomments off}

بروز رسانی موکز تلفن ویپ 3cx و الستیکس ۵ به صورت Command Line - 5.0 out of 5 based on 1 vote

پورت های مورد استفاده در مرکز تلفن ویپ 3cx و الستیکس

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پورت های مورد استفاده در مرکز تلفن ویپ 3cx و الستیکس

The following is a complete list of ports that 3CX Phone System uses in a default installation scenario:

Protocol

Port (Default)

Description

Port Forwarding Required

TCP

5001 or 443

v15: HTTPs port of Web Server.This port can be configured.

Yes – if you intend on using a 3CX client, Bridge Presence, Remote IP Phones from outside your LAN and 3CX WebMeeting functionality.

TCP

5015

V15: This port is used for the online Web-Based installer wizard (NOT 3CX config command line tool) only during the installation process.

Optional - During the installation process when the Web-Based installer is used from external source

UDP & TCP

5060

3CX Phone System (SIP)

Yes – if you intend on using VoIP Providers, WebRTC and Remote Extensions that are NOT using the 3CX Tunnel Protocol

TCP

5061

3CX Phone System (SecureSIP) TLS

Yes – if you intend on using Secure SIP remote extensions

UDP & TCP

5090

3CX Tunnel Protocol Service Listener

Yes -if you intend on using remote extensions using the 3CX Tunnel Protocol (within the 3CX clients for Windows / Android / iOS) or when using the 3CX Session Border Controller

UDP

9000-10999(default)

3CX Media Server (RTP) – WAN audio/video/t38 streams

Yes – if you intend on using remote extensions or a VoIP Provider

TCP & UDP

TCP - 443, 4443

UDP - 48000-65535

3CX WebMeeting Audio and Video. Not to be opened on the PBX. Must instead be opened on the network on which WebMeetings will take place.

Yes – if you intend on using 3CX WebMeeting

پورت های مورد استفاده در مرکز تلفن ویپ 3cx و الستیکس - 5.0 out of 5 based on 1 vote

تنظیم ایمیل سرور Gmail برای ارسال پیامها در مرکز تلفن 3CX

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Follow the steps below to configure your Gmail account as the Mail Server for 3CX Phone System. In this configuration your Gmail account will be used to send email notifications, welcome e-mails and fax messages.

ادامه مطلب: تنظیم ایمیل سرور Gmail برای ارسال پیامها در مرکز تلفن 3CX
تنظیم ایمیل سرور Gmail برای ارسال پیامها در مرکز تلفن 3CX - 5.0 out of 5 based on 1 vote

پاک کردن مرکز تلفن 3CX و الستیکس از سیستم عامل دبیان

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در صورتی که قصد حذف مرکز تلفن 3CX یا الستیکس ۵ را دارید میتوانید از ۲ روش ذیل استفاده نمایید:

 ۱ - حذف مرکز تلفن ویپ  با حفظ نگه داشتن اطلاعات مانند مکالمات تلفنی ظبط شده ، دیتابیس و ... 

این در زمانی به کار می آید که شما بعدا قصد نصب مجدد مرکز تلفن ویپ خود را داشته باشید.

apt-get remove 3cxpbx

 

۲-  حذف کامل مرکز تلفن ویپ الستیکس ۵ یا 3CX :

apt-get remove --purge 3cxpbx

پاک کردن مرکز تلفن 3CX و الستیکس از سیستم عامل دبیان - 5.0 out of 5 based on 1 vote

ریست پسورد ROOT سیستم عامل مرکز تلفن 3cx و الستیکس

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ریست پسورد ROOT سیستم عامل Debian 9 مرکز تلفن 3cx و الستیکس

Start your Debian 9 machine and Press any key to stop the boot process. Press “e” to edit the kernel command line parameters.

Reset root password on Debian 9 Grub

Use “down arrow” key for scrolling down. Go to the line that starts with the word “linux” and use “forward” arrow or press “End” button to go to the end of the line, and then add “init=/bin/bash“.

 Reset root password on Debian 9 Editing Kernel Commands

After you have added the entry, press “Ctrl + x or F10” to boot Debian 9.

Debian will now boot into single user mode, with the root filesystem mounted in “read-only mode“. So, use below command to mount the root file system in “read-write mode“.

mount -o remount /

Reset root password on Debian 9 Mount Root File System

Finally, change the root user password using “passwd” command.

passwd

Reset root password on Debian 9 Reset Root PasswordReboot your system and use the new password we set now for the root user on your system.

 

{rscomments off}

ریست پسورد ROOT سیستم عامل مرکز تلفن 3cx و الستیکس - 5.0 out of 5 based on 1 vote

اصلاح و تغییر شماره تلفن مخاطب در تماسهای ورودی و خروجی Caller ID Reformatting

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اصلاح و تغییر شماره تلفن مخاطب در تماسهای ورودی و خروجی Caller ID Reformatting

در نظر داشته باشید زمانی که با شماره تلفن ثابت شرکت ما تماس تلفنی گرفته میشود ممکن است شماره تماس گیرنده Caller ID به صورت صحیح  نمایان نگردد.

به عنوان مثال شماره تلفن تهران به صورت ۲۱۴۹۲۸۳۰۰۰ نمایش داده شود.

ادامه مطلب: اصلاح و تغییر شماره تلفن مخاطب در تماسهای ورودی و خروجی Caller ID Reformatting
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تنظیمات فایروال میکروتیک در مرکز تلفن ویپ NAT , PAT

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تنظیمات فایروال میکروتیک در مرکز تلفن ویپ NAT , PAT

جهت ایجاد دسترسی از بیرون شبکه به سرور 3cx میبایست پورتهای لازم را از سمت اینترنت به داخل باز نمود. با استفاده از روتر برد میکروتیک mikrotik  میتوان تنظیمات را به صورت ذیل انجام داد.

ادامه مطلب: تنظیمات فایروال میکروتیک در مرکز تلفن ویپ NAT , PAT
تنظیمات فایروال میکروتیک در مرکز تلفن ویپ NAT , PAT - 5.0 out of 5 based on 3 votes

تنظیم مجدد مرکز تلفن الستیکس ۵ و 3CX در لینوکس

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جهت تنظیم مجدد مرکز تلفن ویپ الستیکس ۵ و یا همان  3CX در لینوکس با ید به سیستم عامل لاگین نموده و دستور زیر را وارد نمایید.

با این رستور میتوانید مجددا از ابتدا تنظیمات مرکز تلفن VoIP خود را انجام دهید.

 

sudo /usr/sbin/3CXWizard --cleanup

تنظیم مجدد مرکز تلفن الستیکس ۵ و 3CX در لینوکس - 5.0 out of 5 based on 5 votes

ایجاد صف تماس پیشرفته یا کالسنتر در مرکز تلفن ویپ الستیکس ۵ و 3CX

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ایجاد صف تماس پیشرفته یا کالسنتر در مرکز تلفن ویپ الستیکس ۵ و 3CX

ایجاد یک صف تماس پیشرفته ابتدا وارد پنل مدیریتی مرکز تلفن  3CX یا الستیکس ۵  شویم.

ادامه مطلب: ایجاد صف تماس پیشرفته یا کالسنتر در مرکز تلفن ویپ الستیکس ۵ و 3CX
ایجاد صف تماس پیشرفته یا کالسنتر در مرکز تلفن ویپ الستیکس ۵ و 3CX - 5.0 out of 5 based on 1 vote

تنظیم DTMF برای داخلیهای مرکز تلفن ویپ 3cx 

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تنظیم DTMF برای داخلیهای مرکز تلفن ویپ 3cx 

DTMF یا (dual tone multi frequency)  سیگنالی است که شما با فشردن کلید شماره های دستگاه تلفن خود تولید مینمایید. در آمریکا DTMF  با نام Touchtone شناخته میشود.

3cx elastix voip DTMF setting iran

در یک مکالمه تلفنی VoIP  ، DTMF  به صورت یکی از روشهای زیر ارسال میگردد:

  • in-band: (as a beep)
  • out-of-band: (via SIP or RTP signaling messages)
  • SIP-INFO (is not recommended)

مرکز تلفن VoIP  الستیکس و  3CX  روش های بالا را پشتیبانی میکند.

جهت تنظیم DTMF  به منو EXtention رفته و سپس بر روی شماره تلفن داخلی مورد نظر Edit را می زنیم.

dtmf sip setting in elastix and 3cx
مانند تصویر نمونه در قسمت Phone Provisioning در بخش Network گزینه ی DTMF Mode را به In-Band ، SIP INFO  و یا RFC2833 انتخاب می کنیم.

 

تنظیم DTMF برای داخلیهای مرکز تلفن ویپ 3cx  - 5.0 out of 5 based on 1 vote
کدهای مرکز تلفن ویپ 3CX

کدهای مرکز تلفن ویپ 3CX

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کدهای مرکز تلفن ویپ الستیکس و 3CX

در مرکز تلفن ویپ elastix  و 3cx  با شماره گیری کدهای DTMF میتوان کارهای یسیاری را به شرح ذیل انجام داد.

ادامه مطلب: کدهای مرکز تلفن ویپ 3CX
کدهای مرکز تلفن ویپ 3CX - 5.0 out of 5 based on 2 votes

چگونه تماس از دست رفته در هنگام انتفال Blind Transfer تماس را برگردانیم؟

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چگونه تماس از دست رفته در هنگام انتفال Blind Transfer تماس را برگردانیم؟

مرکز تلفن ویپ 3cx دارای قابلیت برگرداندن تماس از دست رفته در زمان ترنسفر یا انتقال تماس کورکورانه Blind Transfer رادارد.

در زمانی که شما از Blind Transfer استفاده مینمایید و داخلی مورد نظر شما اشغال باشد تماس به صورت اتوماتیک به شما برگردانده خواهد شد.

در این موقع شما میتوانید به تماس گیرنده اطلاع دهید که مخاطب ایشان اشغال است و در صورت تمایل به شماره داخلی دیگری انتقال دهید.

فعال کردن قابلیت برگشت تماس

از قسمت منو سمت راست به Setting بروید و گزینه PBX Settings را انتخاب نمایید:

3cx ippbx settings iran

در کادر مشخص شده کد *3* را وارد نمایید و بر روی OK کلیک نمایید.

Callback on Unsuccessful Blind Transfer in 3CX Phone iran

اکنون تماسهای شما پس از ۲ ثانیه به شما برگشت میشود.

 

 

چگونه تماس از دست رفته در هنگام انتفال Blind Transfer تماس را برگردانیم؟ - 5.0 out of 5 based on 1 vote

نحوه ساخت فایل provisioning برای تنظیم داخلی ها

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نحوه ساخت فایل provisioning برای تنظیم داخلی های مرکز تلفن ویپ 3cx  و الستیکس ۵

روش اول:

نحوه ساخت فایل آماده برای تنظیم داخلی ها:

ابتدا  فایل Template را دانلود نموده وآنرا با نرم افزار notepad باز کرده و محتویات آنرامطابق توضیحات زیر تغییر دهید

ادامه مطلب: نحوه ساخت فایل provisioning برای تنظیم داخلی ها
نحوه ساخت فایل provisioning برای تنظیم داخلی ها - 5.0 out of 5 based on 1 vote
مدیریت وضعیت پاسخگویان در صف مرکز تماس تلفن ویپ 3CX

مدیریت وضعیت پاسخگویان در صف مرکز تماس تلفن ویپ 3CX

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مدیریت وضعیت پاسخگویان در صف مرکز تماس تلفن ویپ 3CX

یکی از قابلیتهای مراکز تماس تلفنی صف تماس یا Call Queues میباشد. پاسخگویان به تماس گیرندگان در صف تماس تلفن اعضا نیز گفته میشود. 

زمانی که با شرکت شما تماس گرفته میشود میتوان تماس گیرندگان را به یک صف هدایت نمود و به ترتیب ورود به صف ، پاسخگویان (اعضا)  ، پاسخگوی تماس آنها باشند.

ادامه مطلب: مدیریت وضعیت پاسخگویان در صف مرکز تماس تلفن ویپ 3CX
مدیریت وضعیت پاسخگویان در صف مرکز تماس تلفن ویپ 3CX - 5.0 out of 5 based on 3 votes

راهنما و مستندات آموزشی مرکز تلفن ویپ 3CX

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3CX کسب دانش حرفه ای و ارتقاء شغلی و کسب درآمد شما را تضمین مینماید.

ادامه مطلب: راهنما و مستندات آموزشی مرکز تلفن ویپ 3CX
راهنما و مستندات آموزشی مرکز تلفن ویپ 3CX - 5.0 out of 5 based on 4 votes
تنظیم و راه اندازی فکس سرور 3CX

تنظیم و راه اندازی فکس سرور 3CX

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تنظیم و راه اندازی فکس سرور 3CX

مرکز تلفن ویپ 3CX  به همراه فکس سرور پیشرفته داخلی ارائه میگردد. فکس سرور 3CX به شما این امکان را میدهد تا بتوانید فکس را با فرمت فایل PDF دریافت کند و به یک داخلی یا چند داخلی ارسال نماید.

ادامه مطلب: تنظیم و راه اندازی فکس سرور 3CX
تنظیم و راه اندازی فکس سرور 3CX - 5.0 out of 5 based on 1 vote

دسترسی به سیستم عامل مرکز تلفن ویپ از کنسول مدیریتی

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دسترسی به سیستم عامل مرکز تلفن ویپ از کنسول مدیریتی

با استفاده از قابلیت دسترسی به ترمینال سیستم عامل دبیان از کنسول مدیریتی 3cx دیگر نیازی به استفاده از برقراری ارتباط از طریق SSH  به سیستم عامل مرکز تلفن ویپ  نیست.

ادامه مطلب: دسترسی به سیستم عامل مرکز تلفن ویپ از کنسول مدیریتی
دسترسی به سیستم عامل مرکز تلفن ویپ از کنسول مدیریتی - 5.0 out of 5 based on 4 votes

تنظیمات شماره گیری dtmf در سیستم تلفنی 3cx

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 تنظیمات شماره گیری dtmf در سیستم تلفنی 3cx

در بعضی مواقع در خطوطی مانند سیپ ترانک شماره گیری از داخل و خارج با مشکل مواجه می شود به این معنی که فرد هنگام تماس با ivr یا تلفن گویا هرچه عدد مورد نظر را شماره گیری می کند سیستم متوجه آن نشده

ادامه مطلب: تنظیمات شماره گیری dtmf در سیستم تلفنی 3cx
تنظیمات شماره گیری dtmf در سیستم تلفنی 3cx - 5.0 out of 5 based on 1 vote

زیر مجموعه ها

نصب و راه اندازی مرکز تلفن ویپ


نصب و راه اندازی مرکز تلفن ویپ در شرکت شما

نصب و راه اندازی مرکز تلفن ویپ در شرکت شما ONPREMISE IPPBX voip
  • مشاوره ، طراحی و آموزش
  • نصب تخصصی و حرفه ای مرکز تلفن ویپ
  • اتصال خطوط تلفن شهری SIP Trunk , E1 خطوط 4 یا 5 رقمی
  • راه اندازی قابلیتهای بروز مرکز تلفنی VoIP
  • ارتباط با سایر نرم افزارها

قیمت ویژه : فقط ۱،۵ میلیون تومان قیمت : ۳ میلیون

مرکز تلفن ویپ در Cloud یا سانترال اینترنتی

مرکز تلفن اینترنتی - سانترال اینترنتی - hosted pbx - cloud pbx - ippbx
  • نصب و راه اندازی رایگان
  • ایمن و پیشرفته ترین مرکز تلفن ویپ آلمان
  • بدون نیاز به سرمایه گزاری و خرید تجهیزات
  • خرید و اتصال سریع خط تلفن شهری
  • تلفن گویا IVR، ضبط مکالمه، گزارشات
  • تلفن داخلی رایگان، فکس و ...

قیمت:  شروع از ۶ هزار تومان

 

پارسه پلاس عضو رسمی اتحادیه مخابرات ، سازمان نظام صنفی رایانه کشور و توزیع کننده انحصاری مراکز تلفن ویپ 3CX آلمان و فنویل در ایران و اولین شرکت ارائه دهنده خدمات مراکز تلفن اینترنتی یا Hosted PBX در ایران میباشد.

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