[Asterisk-Users] T100P Pri Audio

[Donny Kavanagh]

I've been working on an asterisk box at work for a few weeks now, things
were finally starting to sail smoothly until I hit this head scratcher
this morning.

It's a rather intricate problem, so bear with me. Heres the scenario.

What works:
If I call from my sip phone -> sip phone everythings ok
If I call from sip phone -> external pots number ok as well
If I map one of our did's off the pri to one of our internal sip
extensions, it also works perfect.
If someone calls me or I call them in the 3 ways I mentioned above,
music on hold works.
If I call an external did mapped to one of our sip servers, it also
works great.

My config for the extension in question, looks like so (it was much more
then this, but for the sake of simplicity this is enough)

exten => 6261,1,Answer()
exten => 6261,2,Wait(2)
exten => 6261,3,Background(ivrmenu) 

If I call this extension from my sip phone, it will answer, audio will
play and the log looks like so:

    -- Executing Answer("SIP/donnyhome-78be", "") in new stack
    -- Executing Wait("SIP/donnyhome-78be", "2") in new stack
    -- Executing BackGround("SIP/donnyhome-78be", "ivrmenu") in new
stack
    -- Playing 'ivrmenu' (language 'en')

However, if I call this number externally, or dial from the sip phone
and dial with an 8<full number> to force it to go outside and come back
in, I see this.

    -- Executing Dial("SIP/donnyhome-efff", "Zap/g1/85626261") in new
stack
    -- Called g1/85626261
    -- Zap/1-1 is making progress passing it to SIP/donnyhome-efff
    -- Executing Answer("Zap/11-1", "") in new stack
    -- Accepting call from '6135626100' to '6261' on channel 0/11, span
1
    -- Executing Wait("Zap/11-1", "2") in new stack
    -- Executing BackGround("Zap/11-1", "ivrmenu") in new stack
    -- Playing 'ivrmenu' (language 'en')
    -- Channel 0/11, span 1 got hangup
  == Spawn extension (sip, 6261, 3) exited non-zero on 'Zap/11-1'
    -- Hungup 'Zap/11-1'
    -- Channel 0/1, span 1 got hangup
    -- Zap/1-1 is circuit-busy
    -- Hungup 'Zap/1-1'
  == Everyone is busy/congested at this time
Oct  7 18:26:35 WARNING[-188933200]: pbx.c:1933 ast_pbx_run: Timeout,
but no rule 't' in context 'sip'

The audio will start playing according to the cli, however I hear
nothing on the phone outside the pri and shortly there after it hangs
up.

I'm pulling out my hair at this one, I'm still running 1.0.0 so I don't
know if that's the issue, but I wanted to install via rpm's and the fc2
rpms arnt available for 1.0.1 as of yet and I havnt had time to make
them myself.

Any insight into this would be greatly appreciated.