I had a working trunk for Viatalk that seems to be giving me grief now. I have a default any DID/any CID inbound route that routes all inbound calls to a SIP extension on an SPA3102. When an inbound call comes in, the call is immediately sent to voicemail. When I call that extension from either an internal SIP extension or an external SIP device the extension rings normally.
When I turn on SIP debugging, I can see the SIP INVITE request going to the SPA3102 and the SPA3102 responds with a Trying message followed by a Busy Here message. Yet if the call came from an external SIP extension, I see the identical SIP INVITE going to the SPA3102 and again the Trying message coming back from the SPA3102, but this is then followed by a Ringing message as expected.
Any ideas as to why I’m getting the busy back from the SPA3102 or why this might not be working?
That’s a weird one but the first thing that I would look at would be to see if it’s an incompatible codec issue - that is, if VT is sending using “bandwidth saving” codec and Asterisk is passing it through but the SPA3102 either cannot handle that codec, or you have it set not to permit use of that codec.
Second thought… did you perhaps accidentally activate last caller blocking (or whatever it’s called) the last time you called from the number you’re using to test this? This is a feature that can be set at the adapter and I think it may return “Busy Here” if the call is blocked. If someone else calls from a different number, does the call go through?
There’s a couple of things to check… just shooting in the dark though.
Someone must have accidentally activated the last caller block feature. I guess I better disable those things! Deactivating the Block Last feature solved the problem!
Again, thanks so much!