December 7, 2011 – 6:58 pm | 16 Comments

3CX is pleased to announce a new release of 3CX Phone System 10, build 22686. Service Pack 5 fixes numerous 3CX Wallboard issues and also adds a new and important caller ID feature. To download …

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Home » Docs & FAQ

3CX Phone System Parameters Table

Submitted by on January 10, 2010 – 9:01 pmNo Comment

The 3CX Parameters table allows advanced options to be set. We do not recommend changing these options without having been requested to do so by support.

This is a list of the various additional settings available via the parameters table:

ParameterDefault ValueDescription
ALLOWEXTERNALONN/A
ALLOWNOSDPIFBOUNDTOMSN/A
ALLOWSOURCEASOUTBOUND0Default Value "0"
3CX Version >=8

When set to "1", 3CX PhoneSystem will use source IP/port of
SIP request instead of Contact field provided by SIP entity.
Media Server will use source of incoming RTP traffic for
destination of outgoing traffic. This parameter may be set
when external entities cannot provide correct information
for some reason, including (but not limited to):

remote entity is behind double/triple NAT
external SIP device doesn't support NAT traversal (RFC3581)
external SIP device implementation provides inconsistent
information (bugs in implementation)
remote network has a SIP ALG somewhere and the SIP ALG
performs incorrect modifications of SIP messages
Setting this parameter may assist with resolving some issues
with remote extensions or VoIP Providers which have NAT
traversal issues.

NOTE: This is provided for experimental purposes only and
may create unpredictable results with different devices or
network layouts.

ALLOWUSEBUSYOPTIONFORGROUP0Default Value "0"
3CX Version >=8

When set to “1″ this will enable multiple calls to members of a
Ring Group when the extensions of the Ring Group members
have a Busy Detection Mechanism set to ‘Use Phone status’
BMACCEPTTOUT303CX Version >=8

Rebound acceptor timeout in seconds
BMCALLTOUT303CX Version >=8

Rebound caller timeout in seconds
CALLHISTTEMPLPERCALL3CX Version >=8

Full Path to the XML Template file for CDR Output with each
call in a separate file
CALLHISTTEMPLSINGLE3CX Version >=8

Full Path to the XML Template file for CDR Output with all
calls in one file
CALLHISTTEMPLSOCK3CX Version >=8

Full Path to the XML Template file for CDR Output to a Network Socket
CMPORT5482Listening port for Comm Server - Still part of INI file.
CONFPLACE_RTPFIRST403103CX Version >=8

This specifies the first port in the range which the Conference Room Server will use to send and receive audio streams for Conference Calls. The range of ports is set by default to 128 ports, so the Conference Room Server will be able to handle 64 simultaneous calls.
CONFPLACE_RTPLAST404383CX Version >=8

This specifies the last port in the range which the Conference Room Server will use to send and receive audio streams for Conference Calls. The range of ports is set by default to 128 ports, so the Conference Room Server will be able to handle 64 simultaneous calls.
CONFPLACE_SIPPORT403003CX Version >=8

This specifies the SIP Port which the Conference Room Server will use to send and receive call requests. Please note that if this parameter is changed, it will be necessary to restart the "3CX PhoneSystem Conference Room" Service.
DISABLE_SYSCALL_AUTH0Description: This option controls whether INVITE requests
from System Extensions need to be authenticated.
Valid Values:
0 - INVITE from system extensions requires authentication
(default behavior)
1 - INVITE from system extensions does not require
authentication. It excludes additional INVITE transaction from
call flow.

System Extensions are: IVR (Makecall helper),
Queue Manager ("polling" call), and Paging group
(Intercom calls to members of a paging group)
ENABLEEARLYMEDIANA
EXPIRATIONGRACEPERIOD0When a registration expires you may configure a value in
seonds during which the registration shall still be
considered as active.
EXPIRYCHECKTIMENA
FAXDIRECTSDP1When an inbound fax is received from an external entity,
such as a VOIP Provider, then the SDP contains a T38
codec, the PBX will process T38 directly to the 3CX FAX server
extension. This is triggered only when the forwarding
destination is the Fax extension number (ie default 888 in a
3 digit scenario)
FORCEREAUTH0Forces 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
IVR_RTPFIRST406103CX Version >=8

This specifies the first port in the range which the Digital
Receptionist Server will use to send and receive audio streams
for Digital Receptionist and Queue Calls. The range of ports is
set by default to 256 ports, so the Digital Receptionist Server
component will be able to handle 128 simultaneous calls.
IVR_RTPLAST408663CX Version >=8

This specifies the last port in the range which the Digital
Receptionist Server will use to send and receive audio streams
for Conference Calls. The range of ports is set by default to 256
ports, so the Digital Receptionist Server component will be able
to handle 128 simultaneous calls.
IVR_SIPPORT406003CX Version >=8

This specifies the SIP Port which the Digital Receptionist Server will
use to send and receive call requests. Please note that if this
parameter is changed, it will be necessary to restart the
"3CX PhoneSystem Digital Receptionist" Service.
KEEPALIVEENABLED0Allow 3CX to send keep alives to a VoIP provider and is used in
situations were static port mappings are not possible.
LOCALSUBNETSNA
MAXCALLDURATION10800The maximum duration of a single call. Calls exceeding this
duration will be terminated by the PBX.
MAXEXPIRATION1800Upper limit of registration lengths, in seconds. By default this
is not set and any register request with an expiration greater
than 1800 will be replied to with a 1800 seconds expiry. This
entry is only required if a re-registration time of more than
1800 is required.
MAXNOANSWERTIMEOUTNA
MSADDRFC2833FORINBANDDTMFNA
MSEXTERNALINTERFACEBLANKNetwork card to use to communicate with external entities when
STUN is disabled. When STUN is ENABLED this value is
BLANK. When STUN is DISABLED the value should read the
IP address of the NIC with public / internet access.
MSLOCALINTERFACENA
MSMAXCACHEDFILESIZENA
MSMAXTOTALCACHESIZENA
MSRTPTOSThis 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
"DisableUserTOSSetting"
under "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.
MSSTUNADDRNA
NONCEEXPIRATIONDefines nonce expiration time (seconds). By default it is 1 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.
PERS_PHBK_ONLY03CX Version >=8

Implemented in V8. Setting the value to '1' will send ONLY the
Personal phonebook to the provisioned phone. POLYCOM
phone ONLY.
PARKORB_RTPFIRST400103CX Version >=8

This specifies the first port in the range which the Parking
Orbit Server will use to send and receive audio streams for
Conference Calls. The range of ports is set by default to 128
ports, so the Parking Orbit Server will be able to handle 64
simultaneous calls.
PARKORB_RTPLAST401383CX Version >=8

This specifies the last port in the range which the Parking
Orbit Server will use to send and receive audio streams for
Conference Calls. The range of ports is set by default to 128
ports, so the Parking Orbit Server will be able to handle 64
simultaneous calls.
PARKORB_SIPPORT400003CX Version >=8

This specifies the SIP Port which the Parking Orbit Server will use
to send and receive call requests. Please note that if this parameter
is changed, it will be necessary to restart the "3CX PhoneSystem
Parking Orbit" Service.
REPORTSOURCECONFLICTSPBX will add log report if IP/port of SIP request doesn’t equal to
Contact field. Media Server will report if source of RTP stream
is not equal to SDP connection attributes.
RPIDPRIVACYNA
RPIDSCREENNA
SIPDOMAINmysipdomain.localSIP Domain which the PBX is authoritative for. This value may be
used as the host part of the AOR (Address-Of-Record)
SIPINTERFACEIPNA
SIPPINGPERIOD0Time intervall to poll endpoints with SIP INFO messages,
in seconds. 0 - disabled
SIPREALM3CXPhoneSystemString to use in authentication for the "realm" parameter
SIPTHREADPRIORITYPriority of SIP processing thread. Default is 2 (highest). This
option is still part of INI file.
VMDIALOUTENABLED0When set to '0' option 3 of the Voicemail menu will not allow
the creation of an outbound call. Setting the option to '1' will
enable the outbound calling function.
VMPINREQUIRED1Determines whether VM will ask for pin. By default it is 1. Set to
0 to access the VM menu directly(e.g. Hotel extensions).

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