SPA3102 - all circuits busy on outbound calls


#1

I have the latest asterisk installed (freepbx+asterisk now installed, then upgraded)

Core asterisk is 1.6.2.21

I also have a SPA3102 and am trying to use it for external PSTN, and local DECT (via FXS) capability.

So far

  • I can call internally between multiple extensions
  • An inbound external call correctly routes to the designated internal extension (the spa3102’s internal fxs port, but via asterisk)
  • An outbound external call gets routed successfully via a commercial voip provider
  • trying to make the same route instead use the SPA for outbound results ingetting external dialtone on the SPA3102. (registration is ok)

#2

1.6.2.21 is not the latest version. 1.6.2.x is on security fixes only maintenance. The latest 1.6.2.x is 1.6.2.22. The latest release versions are 1.8.8.1 and 10.0.0 (the former is long term stable, the latter on a faster development cycle).

Without debugging output and details of the configuration, there is not much more that can be said.


#3

interestingly if I listen very carefully to an outbound call

  • I get 100ms or so of ringback
  • I hear the outbound dtmf echoing very quietly
  • I then get stutter dial tone - I’d assumed this was from external call… but if I wait longer … say 10s
  • I get the SPA3102 fast busy

So it almost looks like the outbound call was attempted, but failed.


#4

I was able to capture this from the status screen as I was attempting a call (have mangled numbers for privacy)

It looks like the call might be made/ connected but the call is getting dropped quickly . This status gets shown the whole time I’m hearing stutter dial tone in my ear (calling from internal extension) - however, the called number does not ring …

PSTN Line Status
Hook State: Off Line Voltage: 7 (V)
Loop Current: 30.8 (mA) Registration State: Registered
Last Registration At: 1/1/2012 11:31:50 Next Registration In: 2357 s
Last Called VoIP Number: 02380nnnnnn Last Called PSTN Number: 1571
Last VoIP Caller: 02380nnnnnn Last PSTN Caller: , 05601nnnnnn
Last PSTN Disconnect Reason: VoIP Call Ended PSTN Activity Timer: 30000 (ms)
Mapped SIP Port: Call Type: PSTN Gateway Call
VoIP State: Connected
PSTN State: Connected to PSTN VoIP Tone: None
PSTN Tone: None VoIP Peer Name: 02380nnnnnn
PSTN Peer Name: VoIP Peer Number: 02380nnnnnn
PSTN Peer Number: 1471 VoIP Call Encoder: G711a
VoIP Call Decoder: G711a VoIP Call FAX: No
VoIP Call Remote Hold: No VoIP Call Duration: 00:00:05
VoIP Call Packets Sent: 297 VoIP Call Packets Recv: 294
VoIP Call Bytes Sent: 47520 VoIP Call Bytes Recv: 47040
VoIP Call Decode Latency: 50 ms VoIP Call Jitter: 0 ms
VoIP Call Round Trip Delay: Not Available VoIP Call Packets Lost: 0
VoIP Call Packet Error: 0 VoIP Call Mapped RTP Port: 16434 >> 0


#5

double checked again –
a) removing power from SPA3102 (which means direct pstn connect, no voip) and the analogue line had stutter dial tone (messages waiting).
b) cleared messages
c) powered back up / voip registered
d) outbound call now gets plain dial tone

so I think when I’m making an outbound call I hit the spa3102 and get the pstn line dial tone, but I’m not dialling through.

I have the outbond dial plan set to none, the gw is enabled, and one-stage dialling is enabled ??