This article describes how I successfully configured the Sipura SPA-3000 (fw 2.0.13) for use as a single line inbound/outbound trunk within Asterisk at Home (asterisk 1.2.1). Unlike the other examples I found, this configuration is fairly simple and does NOT require configuration of special extensions, etc. This configuration should be fairly secure, but any suggestions and/or feedback are very welcome!
When incoming calls are received by the SPA-3000, they are forwarded to the Asterisk PBX with CALLER ID information and can be routed like any other POTS trunk (ie: as per Incoming Calls config and/or Inbound Routing config by CID). When outgoing calls are placed through the SPA-3000, this device dials the number and connects the call. The person making the call WILL hear the DTMF tones (aka touch tones) that are dialed by the SPA-3000 just before the call is connected. I have not been able to find a way of preventing this (yet).
Configuring Trunk within Asterisk PBX using AMP
Login to AMP (Asterisk Management Portal). Navigate to Setup, Trunks, and choose “Add SIP Trunk”.
General Settings
1.Outbound Caller ID: (leave blank - cannot be used by POTS line)2.Maximum Channels: 1 (required - see note below)NOTE: Each SPA-3000 supports a single channel. You need to setup multiple trunks for multiple SPA-3000 devices.
Outgoing Dial Rules
1.Dial Rules:2.1+NXXNXXXXXX ; prefix 10 digit dialing with "1"3.1NXXNXXXXXX ; allow all 11 digit dialing as-is4.NXXXXXX ; allow all 7 digit dialing as-isOutgoing Settings
01.Trunk Name: pstn_spa0102. 03.Peer Details:04.auth=md505.context=from-pstn06.dtmfmode=inband07.fromuser=asterisk08.host=10.10.10.21 ; IP address of SPA device09.insecure=very10.nat=yes ; omit if no NAT exists between PBX and SPA11.port=506112.secret=01234567890113.type=peer14.username=asteriskIncoming Settings
01.User Context: spa0102. 03.User Details:04.allow=ulaw05.context=from-pstn06.disallow=all07.dtmfmode=inband08.host=10.10.10.21 ; IP address of SPA device09.insecure=very10.nat=yes ; omit if no NAT exists between PBX and SPA11.secret=KzBTALezmG1a12.type=friendRegistration
1.Register String: ; omit - not necessary to register w/ SPA device?Configuring Outbound Routing within Asterisk PBX using AMP
Login to AMP (Asterisk Management Portal). Navigate to Setup, Outbound Routing, and choose “Add Route”.
Add Route
01.Route Name: ; user preference, avoid special characters here?02.pstnspa103. 04.Dial Patterns: ; dial 5 plus 11 digit, 10 digit, and 7 digit numbers05.; omit each "5|" to use trunk without dialing prefix06.5|1NXXNXXXXXX ; accept 5 + 11 digit dialing07.5|NXXNXXXXXX ; accept 5 + 10 digit dialing08.5|NXXXXXX ; accept 5 + 7 digit dialing09. 10.Trunk Sequence: ; add each available SPA-3000 trunk11.SIP/pstn_spa0112.SIP/pstn_spa0213.SIP/pstn_spa03Configuring the Sipura SPA-3000
The following example only illustrates changes to default settings. Start by performing a factory reset of your SPA-3000. Connect a handset to the PHONE jack on the SPA-3000 and dial “****” to access the configuration menu, then dial “73738#” (aka “RESET#”) to perform a factory reset.
Login to the web interface of your SPA-3000, click “Admin”, then click “Advanced”. Configuration changes for each tab/page are shown below.
SYSTEM
01.USER PASSWORD: secretpwd ; secures the SPA web interface02.; username 'user' or 'admin'?03. 04.DHCP: no ; recommend static ip address05.STATIC IP: 10.10.10.2106.NETMASK: 255.255.255.24007.GATEWAY: 10.10.10.3008. 09.HOSTNAME: voip-spa1 ; optional10.DOMAIN: example.net ; optional11.PRIMARY DNS: 10.10.10.2 ; optional12.SECONDARY DNS: 10.10.10.3 ; optional13.PRI NTP: ntp1.example.net ; optional14.SEC NTP: ntp2.example.net ; optionalSIP
1.RTP Packet Size: 0.020 ; improves sound quality (was 0.030)?REGIONAL
1.TIME ZONE: GMT-05:00 ; Central Time ZonePSTN LINE
01.NAT Mapping Enable: yes ; only change if NAT exists between PBX and SPA02.NAT Keep Alive Enable: yes ; only change if NAT exists between PBX and SPA03. 04.PROXY: 10.10.10.24 ; IP address of Asterisk PBX05.USE OUTBOUND PROXY: yes06.REGISTER: no07.REGISTER EXPIRES: 360008.MAKE CALL W/O REG: yes09.ANSW CALL W/O REG: yes10. 11.DISPLAY NAME: ; leave blank12.USER ID: 3501 ; optional?13.PASSWORD: ; leave blank14. 15.DTMF Process INFO: Yes ; default value16.DTMF Process AVT: No ; resolve issues with DTMF17.DTMF Tx Method: Auto ; default value18. 19.DIAL PLAN 8: (S0<:s@10.10.10.24:5060>)20.; forwards incoming PSTN calls to PBX21.; resolve issues with DTMF22. 23.VOIP-TO-PSTN GW ENABLE: yes24.VOIP CALL AUTH METHOD: http digest25.ONE STAGE DIALING: yes26.LINE1 VOIP CALLER DP: none27.VOIP CALLER DEFAULT DP: none28.LINE1 FALLBACK DP: none29. 30.VOIP USER 1 AUTH ID: asterisk31.VOIP USER 1 DP: none32.VOIP USER 1 PASSWORD: 01234567890133. 34.PSTN-TO-VOIP GW ENABLE: yes35.PSTN CALL AUTH METHOD: none36.PSTN RING THRU LINE 1: no ; incoming calls do not ring LINE137.PSTN CID FOR VOIP CID: yes38.PSTN CALLER DEFAULT DP: 839. 40.PSTN ANSWER DELAY: 5 ; answer incoming PSTN call in X sec41.; need to allow time for CALLER ID42.; if no CID, you can safely set to 043.; was set to 16Note regarding FAX transmissions
http://jrklein.com/2006/10/15/configuring-sipura-spa-3000-as-trunk-within-asterisk-voip-pbx-server/
Miyabi
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