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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)
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