SIP to PSTN how?

How do you run your Asterisk?

  • Dedicated Machine
  • VMWare
  • Solaris Zones
  • Xen
  • User Mode Linux
  • Sanctioned Appliance

0 voters

I’ve spent the past couple of days shoehorning Asterisk 1.4.18.1 into an x86 Solaris zone. I finally got it running, and it’s connected to my provider as well as an x-lite client on my workaday windows box. I use Speakeasy for DSL and have their home VOIP service, and a static IP.

Now I don’t know a lot about VOIP - I got into this because the ATA is totally boring. So the problem I’m trying to solve is making calls as I normally do via the ATA to the PSTN universe.

When I try to call from x-lite I get this…

– Executing [12126208600@numberplan-custom-2:1] Macro(“SIP/10-081e1998”, “trunkdial|SIP/trunk_1/12126208600|”) in new stack
– Executing [s@macro-trunkdial:1] Set(“SIP/10-081e1998”, “CALLERID(all)=”) in new stack
– Executing [s@macro-trunkdial:2] Dial(“SIP/10-081e1998”, “SIP/trunk_1/12126208600”) in new stack
– Called trunk_1/12126208600
– Got SIP response 604 “Does not exist anywhere” back from 63.123.133.45
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-trunkdial:3] Goto(“SIP/10-081e1998”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-trunkdial,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp(“SIP/10-081e1998”, “”) in new stack
== Auto fallthrough, channel ‘SIP/10-081e1998’ status is ‘CHANUNAVAIL’

…What do I have to do to have Asterisk or Speakeasy “figure out” what kind of target (SIP/PSTN/other?) I’m after? Or perhaps I don’t understand the problem at all!

Thanks for any advice you can offer.