SPA3102 will call out, but not into Asterisk box from PSTN


I need help configuring web interface of the PSTN side of a SPA3102.

I have a new install of CentOS with Asterisk 1.4.26 installed. The setup consists of a PAP2T-NA with 2 simple POTs phones (John & Beverly) and a SPA3102 connected to the PSTN through the FXO RJ11 and another simple POT (Tasha) is on the FXO side. The WAN side is connected to a switch that connects to the Asterisk box and other computrs in my house. There is nothing connected to the LAN port of the SPA3102 and I have it configured so that I can see the SPA3102 web interface from any computer on the LAN.

All three phones can call eachother, or be called, through the Astrisk box and you can see the activity in the Asterisk CLI.

I can call out from any POTs to the PSTN and it shows on the Asterisk CLI, so the SPA3102 is partly ‘in the loop’.

However, when I call into the SPA3102 from the PSTN, I can hear the line ring on the calling phone (a cell phone or a second line not connected to Asterisk) exactly three times and the call hangs up. On about the second ring, the POTs phone (Tasha) on the FXO side of the SPA3102 starts to ring and displays the correct caller ID information for the calling phone. If I pick up the POTs phone, I will connect to the calling phone and the line will remain open until I hang up either the POTs or the calling phone. During this time, there is no activity on the Asterisk CLI. It seems that Asterisk is not involved in this connection, yet is is aware of the SPA3102. the CLI command: sip show peers gives:
Name/username Host Dyn Nat ACL Port Status
PSTN1/PSTN1 D N 5063 Unmonitored

Tasha/Tasha D 5062 OK (8 ms)

Beverly/Beverly D 5061 OK (14 ms)

John/John D 5060 OK (15 ms)

Where PSTN1 is the FXO of the SPA3102 with Tasha on the FXS and John and Beverly being lines 1 and 2 of the PAP2T-NA.

I expect that I have the SPA3102 missconfigured by now. Could somebody please guide me through configuring the PSTN side of the SPA3102?

For completeness, here are some parts of my .conf files:



;srvlookup=yes ; useful if you have an external SIP provider?

[John] ; Ex. 2001, PAP2T-NAa-1, port 5060


              ; friend, so John can make and take calls

;secret=goodpass ; we are not authenticating yet, later

qualify=yes ; Asterisk will check to see if devise is live

nat=no ; There is no router between Asterisk & PAP2T-NAa-1

host=dynamic ; The firewall will assigne a address, but the PAP2T-NA

              ; must register with Asterisk when it boots

port=5060 ; every phone gets it own port

;canreinvite=yes ; usefule if you are using an external SIP provider?

context=internal ; The device ‘John’ (PAP2T-NAa-1) will be assigned the

              ; extension 2001 and switched to another device under

              ; the context 'internal' in /etc/asterisk/extensions.conf

[Beverly] ; Ex. 2004, PAP2T-NAa-2, port 5061






port=5061 ; every phone gets it own port

context=internal ; The device ‘Beverly’ (PAP2T-NAa-2) will be assigned the

              ; extension 2004 and switched to another device under

              ; the context 'internal' in /etc/asterisk/extensions.conf

[Tasha] ; Ex. 2005, SPA3102a-1, port 5062






port=5062 ; every phone gets it own port

context=internal ; The device ‘Tasha’ (SPA3102a-1) will be assigned the

              ; extension 2005 and switched to another device under

              ; the context 'internal' in /etc/asterisk/extensions.conf

; PSTN phone: Line 1

[PSTN1] ; Ex. 2091, SPA3102a-2, port 5063











port=5063 ; every phone gets it own port



[globals] ; contect global

[general] ; context general defined in iax.conf & sip.conf

autofallthrough=yes ; from * TFOT

[internal] ; context internal defined in iax.conf & sip.conf

exten => s,1,Answer()

exten => s,2,Background(enter-ext-of-person) ; have them input an extension

exten => 133,1,Playback(vm-nobodyavail)

exten => 133,2,Hangup() ; Nobody there, dump them

exten => i,1,Playback(pbx-invalid)

exten => i,2,Goto(incoming,s,1) ; send them back to try again

exten => t,1,Playback(vm-goodbye)

exten => t,2,Hangup ; too slow, no hangup > dump them

; John 2001 SIP PAP2T-NAa-1

exten => 2001,1,Answer() ; Answer the call

exten => 2001,n,Verbose(1|Call to John 2001 SIP PAP2T-NAa-1) ; Log it

exten => 2001,n,Dial(SIP/John,15) ; Ring the extension

exten => 2001,n,Hangup() ; End the call

; Beverly 2004 SIP PAP2T-NAa-2

exten => 2004,1,Answer() ; Answer the call

exten => 2004,n,Verbose(1|Call to Beverly 2004 SIP PAP2T-NAa-2 ) ; Log it

exten => 2004,n,Dial(SIP/Beverly,15) ; Ring the extension

exten => 2004,n,Hangup() ; End the call

; Tasha 2005 SIP SPA3102a-2

exten => 2005,1,Answer() ; Answer the call

exten => 2005,n,Verbose(1|Call to Tasha 2005 SIP SPA3102a-2) ; Log it

exten => 2005,n,Dial(SIP/Tasha,15) ; Ring the extension

exten => 2005,n,Hangup() ; End the call

; experiment with external calling
exten => _NXXNXXXXXX,1,Dial(SIP/PSTN1/${EXTEN}) ; Call an external 10 digit

exten => _NXXNXXXXXX,2,Hangup() ; End the call

; This works, I can call out from any POTs phone

What diagnostis should I run or what other files should I post so that people can help?

Thank you all in advance.

John T