e: Connect back to back 2 SPA-3102

Options 06-22-2008 11:16 PM There are probably several ways you can configure it. I will outline one way that will ring the distant spa on an incoming pstn call and will return a pstn dial tone to the distant spa when the attached handset goes off hook. The distant ata is a SPA3102. As an alternate you could also use a different but similiar Linksys adapter such as a PAP2. On the SPA3102 attached to the PSTN line. PSTN Line Tab Line Enable: YES NAT Mapping Enable: yes or no depending on the router used NAT Keep Alive Enable: yes or no depending on the router used Register: No Make Call Without Reg: Yes Ans Call Without Reg: Yes UserID: spaone VoIP-To-PSTN Gateway Enable: yes VoIP Caller Auth Method: none VoIP Default DP: 2 VoIP Caller Default DP: none none will give you the actual PSTN dial tone when the VoIP-to-PSTN gateway answers the call

PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 2 Dial Plan 2: (S0<:spatwo@ipaddress: 5060> ) The dial plan will automatically dial the distant SPA3102 Line 1 when the PSTN-to-VoIP gateway answers the call Where spatwo is the userid on the Line1 tab of the distant SPA3102 (spatwo) Where ipaddress is the ip address of the distant SPA3102 Where 5060 is the sip port number of the Line 1 tab on the distant SPA3102 VoIP Answer Delay: 0 PSTN Answer Delay: 2 Where 2 seconds is long enough to receive an incoming caller id on the pstn line (if you care about it)

Line-In-Use Voltage: 30 Where 30 is about half way between the on-hook and off-hook voltage of your pstn line (polarity disregarded)

Line 1 Tab Enable IP Dialing: YES On the Distant SPA3102 Line 1 tab Line Enable: Yes NAT Mapping Enable: yes or no depending on your router NAT Keep Alive Enable: yes or no depending on your router Register: No Make Call Without Reg: Yes Ans Call Without Reg: Yes UserID: spatwo DialPlan: (S0<:spaone@ipaddress:5061> ) Where spaone is the useid on the pstn line tab of the SPA3102 attached to the pstn line Where ipaddress is the ipaddress of the SPA3102 attached to the pstn line Where 5061 is the sip port number on the pstn line tab on the SPA3102 attached to the pstn line Enable IP Dialing: YES Note: Depending on your network you may or may not have to have NAT Mapping Enable. If you do have NAT Mapping Enable you may also need to configure a STUN server depending on the router that you are using. If the two ip addresses are not on the same local network you will need to forward the sip port in the router to the SPA3102 and you will need to use the external ip address. Hi all, For a first a customer of mine required a 1KM wireless link to provide PSTN from one office to another, the WiFi has professional MicroNet access points in bridge mode with very low latency < 10ms. With some help from this forum I 'tuned' the two SPA3102's to work as follows. PSTN SPA called 'office' Regional TAB Customise settings to suit Australian conditions with a default dial tone for internal calls FXS port impedance to 600 ohms or Uniden WDECT would not ring always More echo suppression: No Line 1 TAB Line Enable Yes Use Proxy / Register No Answer // Make call without Registration Yes Display name home // ID home Dial string (<##:remote@192.168.0.21:5060>) This dial string will call the remote phone with ## Enable IP dialling Yes PSTN TAB Line Enable Yes Use Proxy / Register No Answer // Make call without Registration Yes Display name office // ID office

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Dial Plan 1: (S0<:remote@192.168.0.21:5060>) VoIP to PSTN Gateway enable Yes Line 1 VoIP Caller DP: 1 Line 1 Fallback DP: 1 *critical* VoIP Call DP: none PSTN to VoIP Setup Gateway enable Yes PSTN ring thru Line 1 No PSTN Default DP: 1 VoIP Answer Delay: 0 PSTN Answer Delay: 2 Line in use voltage 30 If a 'fax duet' is in use it can be ignored with default Ring Validation Time: of 256MS If a 'fax duet' is in use it can ring the remote phone with Ring Validation Time: of < 150MS

Remote SPA called 'remote' Regional TAB FXS port impedance 600 ohms More echo suppression: no Customise settings to suit Australian conditions with a default dial tone for internal calls Line 1 TAB Line Enable Yes Use Proxy No Register No Make call without reg // Ans call without reg: yes Display name remote // ID remote Dial Plan: (<0: office@192.168.0.20:5061>|<##: office@192.168.0.20:5060>) This Dial Plan provides a PSTN dial tone if the caller goes off hook and presses '0' // zero This Dial Plan rings the other phone at the PSTN end as an intercom if the caller goes off hook and presses '##' If a PSTN dial tone is required immediately the Dial Plan is: (<S0: office@192.168.0.20:5061>) Enable IP Dialling Yes The rest of the configuration on the two SPA3102's is default. If in doubt, reset them and try again. When an inbound PSTN call was made, the 'remote' phone took two rings to respond, that is, the caller would hear two rings or more prior to the callee hearing any ring.

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