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 einstellenhttp://wiki.3cx.com/documentation/phone-configuration/timezone-provisioning

•Nebenstelle bearbeiten•Telefon auf Provioning konfigurieren

•DHCP 66 Optionhttp://www.3cx.com/sip-phones/DHCP-option-66.html

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

•Quick Stephttp://10.172.0.150/admin/resync?http://10.172.0.2:5481/provisioning/$MA.xmlhttp://10.172.0.150/admin/resync?http://10.172.0.2/management/provisioning/$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/provisioningPolycomhttp://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 IISAastrahttp://10.0.0.11/management/provisioning/Grandstream10.0.0.11/management/provisioningPolycomhttp://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\Templates

• Bearbeiten– Interface– Editor

• Auslesen der Konfigurationhttp://192.168.1.101/admin/spacfg.xml

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 Konfigurationhttp://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 OKIm 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 Funktion5060

Stun.3cx.com

3347

5061

Stun.3cx.com

3348

Öffentliche IP

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

PBX

3CX Phone SystemOutsideInside

IP PBX:5060 (SIP) Voip-Providers:5060

IP-PBX:5060 (Direct) Remote Phone:Random Port

IP-PBX:5090 (Tunnel) Remote Phone:Random Port

IP-PBX:9000-9049 (RTP) Remote Host:Random Port

NAT TCP/UDP

NAT TCP/UDP

NAT UDP

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 6To enable STUN for remote phone go to page 7

TCP/UDP

Phon

e13CXPhone (tunnel)

• Benifits:– No NAT Outside ->Inside needed– Bandwith saving up to 50%

OutsideInside

Local-IP:Random Port PublicIP-PBX:5090TCP/UDP

Tunnel Out

3CXPhone (direct)• Benifits:

– None• Disatvantage:

– Many 3CXPhones = more NAT Rules

OutsideInside

Local-IP:40.000-40.019 PublicIP-PBX:9000-9049

Local-IP:40.000-40.019 PublicIP-PBX:9000-9049NAT-RTP

Phon

e1

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

UDP

UDP

TCP/UDP

Example 2 3CXPhone (direct)Ph

one2

Phon

e1OutsideInside

Local-IP:40.000-40.019 PublicIP-PBX:9000-9049

Local-IP:40.000-40.019 PublicIP-PBX:9000-9049NAT-RTP

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

Local-IP:40.020-40.039 PublicIP-PBX:9000-9049

Local-IP:40.020-40.039 NAT-RTP

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

PublicIP-PBX:9000-9049

Phone2 should be reconfigured to use different RTP ports then Phone1

UDP

UDP

TCP/UDP

UDP

UDP

TCP/UDP

Example 1 Snom Phone (direct)Ph

one1

OutsideInside

Local-IP: 49152 - 65534 PublicIP-PBX:9000-9049

Local-IP: 49152 - 65534 PublicIP-PBX:9000-9049NAT-RTP

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

UDP

UDP

TCP/UDP

Example 2 Snom Phones (direct)Ph

one2

Phon

e1OutsideInside

Local-IP: 49152 - 57343 PublicIP-PBX:9000-9049

Local-IP: 49152 - 57343 PublicIP-PBX:9000-9049NAT-RTP

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

Local-IP: 57344 - 65534 PublicIP-PBX:9000-9049

Local-IP: 57344 - 65534 NAT-RTP

Local-IP:Local-SIP-Port PublicIP-PBX:5060

RTP Out

SIP Out

PublicIP-PBX:9000-9049

Phone1 should be reconfigured to use a smaller RTP ports rangePhone2 should be reconfigured to use other half of RTP ports of Phone1

UDP

UDP

TCP/UDP

UDP

UDP

TCP/UDP

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 Phone System:

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

• 3CX Phone:

• Snom:

• Cisco/Linksys

– SIP -> NAT Support Parameters

– Ext1 -> Nat Settings

Firewall Log (only for 3CX to show)Light Green are logging events in the firewallNon 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 optionsAnswers 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:

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:

SDSL LinesLine Speed Codec Max Simultaneous Calls1024/1024 kbit/s G.711 101024/1024 kbit/s G.729a 222048/2048 kbit/s G.711 202048/2048 kbit/s G.729a 724096/4096 kbit/s G.711 404096/4096 kbit/s G.729a 1006016/6016 kbit/s G.711 606016/6016 kbit/s G.729a 144ADSL Lines2048/192 G.711 12048/192 G.729a 26017/567 G.711 56017/567 G.729a 12     

Source: QSC AG Germany

For best effect install the firewall betweenthe CPU unit and the wall outlet. Place the jaws of the firewall across thepower cord, and bear down firmly. Be sure to wear rubber gloves whileinstalling the firewall or assign the task to a junior system manager. If thefirewall is installed properly, all the lights on the CPU will turn dark andthe fans will grow quiet. This indicates that the system has entered asecure state. For Internet use install the firewall between the demarc ofthe T1 to the Internet. Place the jaws of the firewall across the T1 linelead, and bear down firmly. When your Internet service provider'snetwork operations center calls to inform you that they have lostconnectivity to your site, the firewall is correctly installed. (© MarcusRanum)