[Asterisk-Users] Sangoma A200 preventing Zap channels
from disconnecting
immediately after PSTN line hangs up (getting empty voicemails)
John Novack
jnovack at stromberg-carlson.org
Mon Apr 24 16:43:47 MST 2006
Mike Garey wrote:
>well, the problem isn't that the card doesn't detect a disconnect,
>it's that it doesn't detect it immediately (or at least within a short period).
>
Odds are that is the telco, and not the Sangoma or Digium card. That is
quite normal for a 10-30 second delay. Not all telco CO's send an
immediate pulse when the caller hangs up.
Is there no way to detect 5-6 seconds of silence by Asterisk?
John Novack
> I'm talking about 10 or so seconds before the channel is
>hung up - which is causing empty voicemail messages to be left when
>the user hangs up before the voicemail starts to record (since the
>channel sticks around, and asterisk thinks the person is still there).
> I tried enabling "busydetect=yes" in zapata.conf, but it didn't make
>a difference.
>
>Mike
>
>On 4/24/06, Mark Phillips <g7ltt at g7ltt.com> wrote:
>
>
>>Likewise here.
>>
>>Using a 10 port FXO card and no problems detecting remote hangup. I'll
>>grant you it can be a little slow sometimes however.
>>
>>On Mon, 2006-04-24 at 16:54 -0500, Rich Adamson wrote:
>>
>>
>>>Mike Garey wrote:
>>>
>>>
>>>>As far as I can tell, after discussing this matter with other asterisk
>>>>users in my area, my telco _does_ provide disconnect supervision.. It
>>>>seems that the problem is actually related to the Sangoma A200 card
>>>>I'm using, as two other people both using this same card have
>>>>expressed the same problem.. Are there any other users on this list
>>>>using the Sangoma A200 FXO port card, and experiencing problems with
>>>>asterisk not detecting when a channel has been disconnected? Thanks,
>>>>
>>>>
>>>Hasn't been a problem here with either the TDM400 or A200D cards (both
>>>are in use in same box).
>>>
>>>Just tested it again from an external pstn phone, calling into asterisk.
>>>When the pstn phone hangs up, asterisk recognized it and dropped the sip
>>>session that was handling the call (to a 7960).
>>>
>>>
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