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voice:ata:spa3102 [2017/04/11 08:12]
gcooper
voice:ata:spa3102 [2021/10/27 12:15] (current)
gcooper
Line 1: Line 1:
-====== Cisco/Linksys SPA-3102 ATA Connection Notes ======+====== Cisco/Linksys SPA3102 ATA Connection Notes ======
  
 FIXME This page is copied from another and is incorrect and incomplete. FIXME This page is copied from another and is incorrect and incomplete.
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 See also: **[[voice:ata:spa2100|Linksys/Sipura SPA-2100 ATA Connection Notes]]** See also: **[[voice:ata:spa2100|Linksys/Sipura SPA-2100 ATA Connection Notes]]**
  
-See also: **[[voice:gateway:grandstream_ata|Grandstream Analog Gateways and ATAs]]**+See also: **[[voice:ata:grandstream_ata|Grandstream Analog Gateways and ATAs]]**
  
-See also: **[[quick_guide:freepbx_ata|FreePBX with Analog Telephony Adapters]]**+See also: **[[voice:pbx:freepbx:freepbx_ata|FreePBX with Analog Telephony Adapters]]**
  
 ===== Seven Digit Dialing ===== ===== Seven Digit Dialing =====
 +
 +{{ :voice:ata:spa3102.jpg?300|SPA3102}}
  
 If you use a SIP trunk provider that requires 10 digits, modify the Dialplan in the Line1 (or Line2) settings by inserting ''<:520>'' to the 7-digit section: If you use a SIP trunk provider that requires 10 digits, modify the Dialplan in the Line1 (or Line2) settings by inserting ''<:520>'' to the 7-digit section:
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 For maximum bandwidth savings, use ''g.729a'' Otherwise, choose (the default) ''g.711u'' (uncompressed) for best audio quality.  It's roughly 24Kbps vs. 92Kbps. For maximum bandwidth savings, use ''g.729a'' Otherwise, choose (the default) ''g.711u'' (uncompressed) for best audio quality.  It's roughly 24Kbps vs. 92Kbps.
  
-Also, be sure to set the ''RTP Packet Size'' to ''0.020'' on the SIP tab.+:!: Be sure to set the ''RTP Packet Size'' to ''0.020'' on the SIP tab. 
 + 
 +===== Paging ===== 
 + 
 +In our limited experience, paging amplifiers are usually connected to an FXO port of a PBX.  **The SPA3102 includes an FXO port**. 
 + 
 +http://wiki.freepbx.org/display/FOP/Using+an+SPA3102+FXO+Port+to+connect+with+Valcom+Paging+units 
 + 
 +==== Overview ==== 
 + 
 +  - Configure a SIP Trunk and Outbound Route in FreePBX 
 +  - Configure the SPA3102 
 +  - Determine "Line-In-Use Voltage" of the FXO port and adjust it to match 
 +    - Otherwise inbound calls in will result in a busy signal (SIP response 503 "Service Unavailable"
 + 
 + 
 + 
 +===== Enable the Web Interface on WAN ===== 
 + 
 +{{ :voice:ata:spa3102_connections.jpg?300|SPA3102 Connections}} 
 + 
 +Whereas the **WAN interface is used to connect to the SIP trunk provider**, the **web management of the SPA3102 is only available on the LAN interface** of the unit.  If your SPA3102 is completely behind another firewall, you will find it more convenient to enable the web management on the WAN interface. 
 + 
 +:!: Do **not** make the web interface accessible if your WAN interface is internet-accessible. 
 + 
 +The default settings will leave the SPA3102 in an un-managable state if you only connect the WAN port to your internal network...a common configuration.  One would have to connect a portable PC to the LAN interface to administer the unit. 
 + 
 +These steps will enable the web interface on the WAN port: 
 +  
 +  - Plug in an analog phone or 'butt set' into the Phone port 
 +  - Dial ''****'' (four asterisks) for the (audible) main menu 
 +  - Dial ''110#'' for the IP submenu 
 +  - Dial ''7932#'' then 
 +  - Dial ''1#'' to confirm
voice/ata/spa3102.1491919951.txt.gz · Last modified: 2017/04/11 08:12 by gcooper