Channel not being released

Hi,

We have installed a new server with :
PBX Firmware: 4.211.64-4
PBX Service Pack: 1.0.0.0

We have a Digium AEX410 card with four FXO modules.

We seam to have an issue with channels being held up after the call is released.

On the PBX Status UI we see
Total Active Call 0
Total Active Channels 1 (Some times more)

If we call in we see the call goes though the time condition and is answered by the IVR. If we then release the call from the calling side the call seams to carry on and times out in the IVR and is sent to our default ring group (as setup in the IVR). When the call is answered it’s just dead noise.

Also if we make some out going calls and then release them and try some further calls we get a prompts saying “all circuits are busy”. If we wait some time and try again the calls will connect.

Does anyone have any ideas ?

Thanks for your support.

Update:

We just made an outgoing call and released it.

[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/601-000001db”, “hangupcall,”) in new stack
[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/601-000001db”, “1?theend”) in new stack
[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/601-000001db”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/601-000001db”, “”) in new stack
[2013-08-12 13:40:34] VERBOSE[22954] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/601-000001db’ in macro ‘hangupcall’
[2013-08-12 13:40:34] VERBOSE[22954] features.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/601-000001db’
[2013-08-12 13:40:34] VERBOSE[22954] sig_analog.c: – Hanging up on ‘DAHDI/2-1’
[2013-08-12 13:40:34] VERBOSE[22954] chan_dahdi.c: – Hungup ‘DAHDI/2-1’
[2013-08-12 13:40:34] VERBOSE[22954] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/601-000001db’ in macro ‘dialout-trunk’
[2013-08-12 13:40:34] VERBOSE[22954] pbx.c: == Spawn extension (outbound-allroutes, 907785902251, 7) exited non-zero on ‘SIP/601-000001db’
[2013-08-12 13:40:34] VERBOSE[22955] app_mixmonitor.c: == MixMonitor close filestream (mixed)
[2013-08-12 13:40:34] VERBOSE[22955] app_mixmonitor.c: == End MixMonitor Recording SIP/601-000001db
[2013-08-12 13:40:34] VERBOSE[2042] sig_analog.c: == Starting post polarity CID detection on channel 2
[2013-08-12 13:40:34] VERBOSE[23008] sig_analog.c: – Starting simple switch on ‘DAHDI/2-1’
[2013-08-12 13:40:48] NOTICE[23010] manager.c: Seems to have passed…
[2013-08-12 13:40:48] WARNING[23008] sig_analog.c: CID timed out waiting for ring. Exiting simple switch
[2013-08-12 13:40:48] VERBOSE[23008] sig_analog.c: – Hanging up on ‘DAHDI/2-1’
[2013-08-12 13:40:48] VERBOSE[23008] chan_dahdi.c: – Hungup ‘DAHDI/2-1’

Does this look correct?
Why is look for CID when a call is releasing?

I’d venture that there’s no disconnect supervision on those analog lines. Is that something your telco provides?

We are using BT in the UK.
I think they use polarity reversal on answer and polarity reversal on release.

What should we have in :

answeronpolarityswitch=???
hanguponpolarityswitch=???

Thanks

Hi,

I have checked the BT spec and I think I was wrong above.
It looks like they break the line when the network releases the call.
Something between 80ms and 1000ms.

Is this what the AXE410 wants?

Andy

No clue, our Support guys may have a better idea. www.digium.com/support

Cheers

Hi Malcolm,

We found something on the web stating the BT’s Disconnect Clear Time was low.
We checked with BT and it was 400ms they have changed this to 800ms and we thought this had cleared our issue but it would seem not.

We are using an AEX410 with four FXO modules. When we purchased the AEX410 it had 3 FXO modules and 1 FXS. We moved an old FXO module we had in TDM400 card.
It would seem the hanging calls are on this port.

The FXO module we moved in is an older Rev than the other three.
Does the AEX410 have a min FXO module Rev?
Can we mix Rev’s of FXO modules in one AEX410?

Thanks
ANdy

Howdy,

“Does the AEX410 have a min FXO module Rev?
Can we mix Rev’s of FXO modules in one AEX410?”

It shouldn’t; but it’s been a while since we sold the TDM400, so perhaps that module’s simply gone wonky with the passage of time. If you move the line from the line+module that’s not working to a different module, and simultaneously put a different line on the possibly wonky module, does the problem follow the line or the module?