3CX Partner Schulung

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3CX VoIP Partner PBX Voice PCX

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3CX Partner Schulung

Connect. Communicate. Collaborate

Installation 3CX Phone
System V8
• Betriebssysteme
•Cassini bis 40 User oder SBS
•Firewall Einstellungen
•Grundeinrichtung
•Nebenstellen
•Provisioning

Provisioning
Cisco SPA

3 Schritte zum
Provisioning
•Zeitzone einstellen
http://wiki.3cx.com/documentation/phone-configuration/timezone-provis
ioning

•Nebenstelle bearbeiten
•Telefon auf Provioning konfigurieren
•DHCP 66 Option
http://www.3cx.com/sip-phones/DHCP-option-66.html

•Manuelles Provisioning
http://www.3cx.com/sip-phones/Cisco-SPA.html

•Quick Step
http://10.172.0.150/admin/resync?http://10.172.0.2:5481/provisioning/$MA.xm
l
http://10.172.0.150/admin/resync?http://10.172.0.2/management/provisionin
g/$MA.xml

Zeitzone Cisco SPA für DE

Provisioning anderer
Telefone




3CX PhoneSystem with Cassini
Aastrahttp://10.0.0.11:5481/provisioning/
Grandstream10.0.0.11:5481/provisioning
Polycomhttp://10.0.0.11:5481/provisioning/
Snomhttp://10.0.0.11:5481/provisioning/cfg{mac}
Ciscohttp://10.0.0.11:5481/provisioning/$MA.xml
 
3CX PhoneSystem with IIS
Aastrahttp://10.0.0.11/management/provisioning/
Grandstream10.0.0.11/management/provisioning
Polycomhttp://10.0.0.11/management/provisioning/
Snomhttp://10.0.0.11/management/provisioning/cfg{mac}
Ciscohttp://10.0.0.11/management/provisioning/$MA.xml

Templates-Vorlagen
• Speicherort (Vista,XP,2008)
C:\ProgramData\3CX\Data\Http\Templ
ates
• Bearbeiten
– Interface
– Editor

• Auslesen der Konfiguration
http://192.168.1.101/admin/spacfg.x
ml

Templates-Speicherort
• Windows XP oder Windows 2003
C:\Documents and Settings\All Users\Application
Data\3CX\Data\Http\Interface\provisioning

• Windows Vista, Windows 7 oder Windows
2008
C:\ProgramData\3CX\Data\Http\Interface\provisioning

Fehlersuche
• Manuelles ansperchen der Konfiguration
http://10.0.0.11:5481/provisioning/$MA.xml
• Syslog des Telefones
– Kiwi Syslog
– Telefoneinstellung (Voice -> System)

SIP Protokoll

SIP Register

SIP Invite (SIP)

Telefon 100 meldet der PBX,
dass es mit der Identität 101
in der Domain
@10.172.0.141 sprechen
möchte. Im Contact definiert
er auf welcher IP:Port es auf
weitere Instruktionen wartet
PBX nimmt die
Informationen an und richtet
einen Invite and die Identität
101 und teilt Ihm mit, dass
es auf der IP:Port auf
Instruktionen wartet.

Im SIP Invite werden keine direkten
Beziehungen der Teilnehmer hergestellt

SIP Invite (SDP)
Im SDP definiert das Telefon 100
auf welchem IP:Port es gerne
Audio erhalten möchte.

SDP OK
Im SDP OK schickt nun die
Identität 101 auf welchen IP:Port
Sie Audio empfangen möchte.
Nach dem ACK kann gesprochen
werden

Log Message 32Sek Voip Call: No ACK
Recieved

Interner Anruf(EXT->EXT)
Contact SDP ist in beiden invites an
101 gleich!
PBX überträgt kein Audio.

SIP/SDP Informationen

RTP Informationen

STUN Funktion
5060

3347
Stun.3cx.com
Öffentliche IP

5061

3348
Stun.3cx.com
Öffentliche IP

Intern an Extern

(Voip oder HomeOffice)

SIP Contact mit STUN

SDP Contact mit STUN

Ports und NAT with 3CX
Anbindung externer
Nebenstellen an der 3CX
Phone System
By Stefan Walther

Vorraussetzungen
• Internet (bevorzugt feste IP for PBX or
DnyDNS.org)

• PBX must have NAT setup accordinly
(Page 3)
• EXT must be bound to Media-Server
• EXT phone must have Stun or Tunnel
activated
– In Stun NAT should be set up

• Followring examples based on
default values

3CX Phone System
Inside Outside
IP PBX:5060 (SIP)
PBX

IP-PBX:5090 (Tunnel)
IP-PBX:5060 (Direct)
IP-PBX:9000-9049
(RTP)

TCP/UDP

NAT TCP/UDP

NAT TCP/UDP

NAT UDP

VoipProviders:5060
Remote Phone:Random
Port
Remote Phone:Random
Port
Remote Host:Random
Port

Assuming default Ports in the PBX and remote phones.
To set fixed ports for remote phones in RTP and SIP go to
page 5 and 6
To enable STUN for remote phone go to page 7

3CXPhone (tunnel)
• Benifits:
– No NAT Outside ->Inside needed
– Bandwith saving up to 50%
Inside Outside
Phone1

Local-IP:Random Port

Tunnel OutPublicIPTCP/UDP
PBX:5090

3CXPhone (direct)


Benifits:
– None
Disatvantage:
– Many 3CXPhones = more NAT Rules
Inside Outside
Local-IP:40.00040.019
Local-IP:40.00040.019
Local-IP:Local-SIP-Port

Phone1



RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049
UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

Example 2 3CXPhone
(direct)
Inside Outside
RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049

Local-IP:40.02040.039
Local-IP:40.02040.039
Local-IP:Local-SIP-Port

RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049

Phone2

Phone1

Local-IP:40.00040.019
Local-IP:40.00040.019
Local-IP:Local-SIP-Port

UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

Phone2 should be reconfigured to use different RTP ports
then Phone1

Example 1 Snom Phone
(direct)
Inside Outside
Phone1

Local-IP: 49152 65534
Local-IP: 49152 65534
Local-IP:Local-SIP-Port

RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049
UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

Example 2 Snom Phones
(direct)
Inside Outside
RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049

Local-IP: 57344 65534
Local-IP: 57344 65534
Local-IP:Local-SIP-Port

RTP Out PublicIP-PBX:9000UDP
9049
NAT-RTP PublicIP-PBX:9000-9049

Phone2

Phone1

Local-IP: 49152 57343
Local-IP: 49152 57343
Local-IP:Local-SIP-Port

UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

UDP

SIP Out
TCP/UDP

PublicIP-PBX:5060

Phone1 should be reconfigured to use a smaller RTP ports
range
Phone2 should be reconfigured to use other half of RTP ports
of Phone1

SIP/SDP Local Port













3CX Phone System:
– Default: 5060
– Configurable: Settings -> Network -> Ports -> SIP Port
3CX Phone:
– Default: Random local Port (somewhere arround 5930)
– Configurable: Connection settings -> advanced settings -> lokal port
Snom:
– Default: Random local Port (somewhere arround 2040)
– Configurable: PhoneGUI -> Advanced -> SIP/RTP -> Network identity (port)
Cisco/Linksys
– Default: 5060 -5090
– Configurable: PhoneGUI (Admin/Advanced) -> SIP -> SIP Parameters -> SIP TCP Port Min/Max
Aastra
Polycom
Grandstram
Yealink
– Default: 5060
– Configurable: PhoneGUI -> Account -> Advanced -> Local SIP Port

RTP Local Port










3CX




Phone System:
Default: 9000-9049 Extern
Default: 7000-7049 Intern
Configurable: Settings -> Network -> Ports -> Ports to use for external leg of Voip provider
Calls
3CX Phone:
– Default: Random local Port between 40000 and 40019
– Configurable: Connection settings -> advanced settings -> RTP-Ports
Snom:
– Default: Random local Port between 49152 - 65534
– Configurable: PhoneGUI -> Advanced -> SIP/RTP -> Dynamic RTP port start /stop
Cisco/Linksys
– Default: Random local Port between 16384- 16538
– Configurable: PhoneGUI (Admin/Advanced) -> SIP -> RTP Parameters -> RTP Port Min/Max
Yealink
– Default: Random local Port between 11780 - 11800
– Configurable: PhoneGUI -> Network -> Advanced -> Local RTP Port

Enable STUN




3CX


3CX



Snom:



Cisco/Linksys
– SIP -> NAT Support Parameters



Phone System:
Default: 9000-9049
Configurable: Settings -> Network -> Ports -> Ports to use for external leg of Voip provider Calls
Phone:

Ext1 -> Nat Settings

Firewall Log (only for 3CX to
show)
Light Green are logging events in the firewall
Non light green events u will not see in the firewall, due to the
connection is already established

Basic messages sent in the SIP environment
•INVITE – connection establishing request
•ACK – acknowledgement of INVITE by the final message receiver
•BYE – connection termination
•CANCEL – termination of non-established connection
•REGISTER – UA registration in SIP proxy
•OPTIONS – inquiry of server options
Answers to SIP messages are in the digital format like in the http protocol. Here are the most important ones:
•1XX – information messages (100 – trying, 180 – ringing, 183 – progress)
•2XX – successful request completion (200 – OK)
•3XX – call forwarding, the inquiry should be directed elsewhere (302 – temporarily moved, 305 – use proxy)
•4XX – error (403 – forbidden)
•5XX – server error (500 – Server Internal Error, 501 – not implemented)
•6XX – global failure (606 – Not Acceptable)
Connection establishing and terminating procedures in the SIP proxy server environment:

Source: QSC AG Germany

SDSL Lines
Line Speed
1024/1024 kbit/s
1024/1024 kbit/s
2048/2048 kbit/s
2048/2048 kbit/s
4096/4096 kbit/s
4096/4096 kbit/s
6016/6016 kbit/s
6016/6016 kbit/s
ADSL Lines
2048/192
2048/192
6017/567
6017/567

Codec
G.711
G.729a
G.711
G.729a
G.711
G.729a
G.711
G.729a

Max Simultaneous Calls
10
22
20
72
40
100
60
144

G.711
G.729a
G.711
G.729a

1
2
5
12

For best effect install the firewall between
the CPU unit and the wall outlet. Place the jaws of the firewall across
the
power cord, and bear down firmly. Be sure to wear rubber gloves
while
installing the firewall or assign the task to a junior system manager. If
the
firewall is installed properly, all the lights on the CPU will turn dark
and
the fans will grow quiet. This indicates that the system has entered a
secure state. For Internet use install the firewall between the demarc
of
the T1 to the Internet. Place the jaws of the firewall across the T1 line
lead, and bear down firmly. When your Internet service provider's
network operations center calls to inform you that they have lost
connectivity to your site, the firewall is correctly installed. (© Marcus
Ranum)

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