• 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
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
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: