ajp26 at buffalo.edu
Mon Oct 29 15:45:49 EDT 2012
>The best way to trouble shoot this is to look at the asterisk log on your server.
>You can follow the call as it is processed by your server and see if it is getting handed to the upstream provider properly or not.
>There are literally dozens of places for thing to go wrong, and most of them will play the same error message to the caller.
I didn't see an answer to this, but, by default there appears to be a context called "radio" indicated in the default node stanza in rpt.conf . That runs through a block in extensions.conf that does some loose checking if a number to dial is valid, and then presents it to pstn-out. Which, by default, btw, only plays an obnoxious congestion tone.
However, numbers that are rejected will route to the "invalidnum" context which plays the "not in service" message. There are a number of predefined ones but, I would guess you are running afoul of toll free numbers, or some non-standard number which is dropping through to one of those patterns.
The asterisk messages log, in /var/log/asterisk/messages , may contain details. You can also watch the console with a bit of verbosity and see what it's trying to do, and see the dialed pattern.
I got this working fairly easily by just adding a SIP peer, but, there's no auth code or anything built in to prevent you from dialing out, or anyone for that matter. For the most part, autopatches are nonexistent over here so nobody knows it is there, I could see someone dialing it accidentally if they're not careful. Dialtone is played over the web transceiver if you patch from there but not the local radio, on my node anyways.
The only thing that's being obnoxious is that it eats DTMF, and for whatever reason, speech seems to be recognized as DTMF sometimes which results in what sounds like choppy conversation, but, it's just that masking function.
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