3CX IP PBX Phone System Technical Training Deerfield.com



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Transcription:

3CX IP PBX Phone System Technical Training Deerfield.com 1 Copyright 2002 ACNielsen

Agenda High level overview of SIP & RTP Call set-up, ports needed and codecs SIP phone configuration VOIP Gateway configuration Server status Log Partner support procedures 2

What is 3CX Phone System? IP PBX based on the SIP standard Works with popular SIP phones, VOIP providers and VOIP Gateways Includes: a SIP server for call set-up a media server for managing audio streams an IVR server for Auto Attendant and Voice Mail Web server (apache) Database server (Postgres) Runs on Windows XP / Vista / 2000 / 2003 3

Benefits of 3CX Phone System No need for separate phone wiring Scalable Add PSTN lines by adding a VOIP Gateway or VOIP Provider Add extensions by plugging in additional phones Extension tied to phone, not network point enables roaming and teleworking 4

SIP Standard What is it? Session Initiation Protocol Text based - similar to the SMTP/HTTP protocol Sets up phone calls only Supported by all major equipment & software manufacturers - mix and match IP software & hardware from different vendors (Cisco, Linksys, Patton Grandstream, etc) Defined in RFC 3261 http://tools.ietf.org/html/rfc3261 5

What is RTP? Real Time Transport Protocol Defines format for delivering audio and video over the internet Each call requires 2 RTP channels, one for each phone, aka Endpoint Defined in RFC 1889 http://tools.ietf.org/html/rfc1889 6

Common SIP Requests REGISTER to register a phone or line with a SIP server INVITE to set-up a call CANCEL to cancel a call set-up BYE to terminate a call 7

Common SIP responses 100 trying 180 ringing 200 OK 401 not authorized 404 destination not found 486 busy Note similarity to HTTP responses! More detail about SIP in technical manual at: http://www.3cx.com/downloads/techmanual31.pdf 8

Typical Call setup Internal call Call setup for a call to another extension or via a VOIP gateway Port range for RTP is 7000- and up (configurable) 2 Ports per Endpoint, one for RTP and one for RTCP! Audio Channel 1 - RTP PORT 7000 Audio Channel 1 - RTCP PORT 7001 Endpoints (phones) communicate with SIP server (3CX IP PBX) via SIP protocol PORT 5060 Audio Channel 2 - RTP PORT 7001 Audio Channel 2 - RTCP PORT 7002 9

Typical Call setup External call Call setup for a call via a VOIP provider or an external extension Port range for RTP to VOIP provider is 9000- and up (configurable) 2 Ports per endpoint, one for RTP and one for RTCP! STUN request needed to get external IP SIP phone connections Audio Channel 1 - RTP PORT 7000 Audio Channel 1 - RTCP PORT 7001 Endpoints (phones) communicate with SIP server (3CX IP PBX) via SIP protocol PORT 5060 Get External IP via STUN request to STUN server PORT 3478 Audio Channel 2 - RTP PORT 9000 Audio Channel 2 - RTCP PORT 9001 Connections to VOIP provider 10

Codecs Codecs encode/compress the audio stream 3CX supports G711 ulaw, mlaw (approx 80 kb per second) GSM (approx 20-30 kb per second) Speex (approx 20-30 kb per second) We do not support G729 (does not support in-band DTMF) G723 Both have high licensing costs, and several other large market players (such as Microsoft) are not including these codecs in their products 11

SIP phone, Gateway configuration Configuring SIP phones & VOIP Gateways 12 Copyright 2002 ACNielsen

SIP phones 3CX supports both hardware and software SIP phones from all popular vendors In theory, we should work with all SIP phones, in practice its better to use tested phones. Supported phones are listed here: http://www.3cx.com/support/sip-phones.html 13

SIP phone configuration Create extension in 3CX Management Console Take note of these details Extension number Authentication ID Authentication Password FQDN / IP of 3CX Phone System Configure these details on the phone 14

Example: SNOM 360 Extension No = Account Password = Authentication Password Registrar = IP of 3CX Phone System Authentication Username = Authentication ID Each vendor has its own terminology! 15

VOIP Gateways VOIP gateways aka FXO gateways convert PSTN lines to SIP extensions Each line on a VOIP Gateway is a SIP extension configuration is similar Gateway is separate network device with own IP It can be anywhere on your network Supported gateways are listed here: http://www.3cx.com/support/voip-gateways.html 16

VOIP Gateway configuration overview Each line on a VOIP Gateway is associated with a virtual extension on 3CX Phone System: PSTN Lines Line 1 Line 2 Line 3 Register individually as virtual extension numbers: 10000 10001 10002 Line 4 10003 VOIP Gateway IP PBX 17

VOIP Gateway configuration To create PSTN lines, click Add PSTN in 3CX Management console. When you click finish, the following details will be provided: IP of 3CX Phone System Virtual extension number for each line Authentication ID for each line Authentication Password for each line Enter these into the Gateway Configuration guides for popular gateways: http://www.3cx.com/voip-gateways/index.html 18

Example: Grandstream GXW 4104/8 Sip user ID, Auth ID, Auth Pw inserted in channels page 19

Check registration in Line Status screen Switch to Line Status in 3CX Management console to see if all lines registered OK 20

VOIP providers 3CX works with mainstream SIP VOIP providers: http://www.3cx.com/support/voip-providers.html Only G711, GSM, Speex codecs are supported We will test against new main stream VOIP providers give us authentication details and 1-2 weeks and we will provide results, with reason of failure if not compatible / recommended. 21

3CX Troubleshooting 22 Copyright 2002 ACNielsen

Most common problems Internet-Facing Firewall configuration Port 5060 open for SIP Port 3478 open for STUN Port 9000- open for RTP (if using VOIP Provider) Registration failures Use of wrong authentication ID Password Extension number not inserted in right field Use of unsupported phones, providers or gateways Use server status log to analyze problem 23

Server status log 3CX Management Console : Phone System > Server status 24

Server status log (2) Logs important messages and errors Call(C:B1): Call from Ln:10500@GW_MX1402_BRI to 999 has been terminated 13:30:35.147 DTMF (in-band) from 192.168.1.11:4930 detected. 13:30:23.935 Call(C:B1): Created audio channel for Ln:10500@GW_MX1402_BRI (192.168.1.11:4930) with Media Server (192.168.1.3:7292) 13:30:08.828 Call(C:B1): got response from 109 13:30:08.703 Call(C:B1): Incoming call from Ln:10500@GW_MX1402_BRI to sip:10500@192.168.1.3 In this example: 192.168.3 is SIP server, GW_MX1402_BRI is gateway with IP 192.068.1.11. Call with ID B1 has been received on virtual line 10500, routed to extension 109, then after the default 15-second timeout, sent to voice mail by the gateway from port 4930 to the media server on port 7292 (meaning the extension is internal), because no answer was received from extension 109. 25

Server status log (3) Server status log shows IP of SIP server, SIP phones, gateways and VOIP providers Shows on which ports audio channels are established (e.g. 192.168.1.3:7292) Shows whether extensions are on same subnet Gives reasons if phone, provider or gateway fail to register Allows you to more easily troubleshoot network and firewall configuration issues Always include server status log with support request! 26

Technical manual More detail about server status log messages in the technical manual at http://www.3cx.com/downloads/techmanualv31.pdf 27

Debug logs Problems during setup, review C:\Program Files\3CX PhoneSystem\install.log Problems with database access C:\Program Files\3CX PhoneSystem\Data\DB\pg_log\*.log Problems with apache web server C:\Program Files\3CX PhoneSystem\Bin\Apache\logs\*.log Problems with SIP server C:\Program Files\3CX PhoneSystem\Data\Logs\3CXPhoneSystem*.log Problems with Media server C:\Program Files\3CX PhoneSystem\Data\Logs\3CXMediaServer*.log 28

Debug logs (2) VoiceMail: C:\Program Files\3CXPhoneSystem\Data\Logs\3CXVoiceBoxManager*.lo g IVR: C:\Program Files\3CX PhoneSystem\Data\Logs\3CXIvrServer*.log C:\Program Files\3CX PhoneSystem\Data\Logs\IvrPhp.log After each phone system service restart, logs are backed up to C:\Program Files\3CX PhoneSystem\Data\Logs\Backup 29

3CX Phone System Versions Free edition Up to 8 configured lines, no support Small Business Edition Up to 16 configured lines and 25 extensions Full version of Call Assistant (call transfer, call park & pickup, etc.) & Message Waiting Indicator (MWI) Ability to buy support and maintenance Price $350 30

3CX Phone System Versions Pro Edition Up to 32 configured lines, UNL extensions Advanced call assistant & MWI Price $895 Enterprise Edition Unlimited extensions and lines Exchange Server 2007 integration Call Queuing Cluster ready Price $1250 31

3CX Support Procedures 32 Copyright 2002 ACNielsen

Support procedures Resellers receive priority technical support to assist in servicing your customers quickly and efficiently Login to your account at https://shop.deerfield.com/resellers/ Submit support incident with SIP phones, VOIP Provider and VOIP Gateways used Detailed problem description, eg: transfer fails when receiving call from VOIP Provider XYZ Include Server Status log Reseller support available 9:00AM to 5:00PM EST 33

Thank you Thank you for attending this technical presentation For questions, please don t hesitate to email us resellersupport@deerfield.com 34