[Freeswitch-users] Problem with freeswith and a Digium card

Alex Crow acrow at integrafin.co.uk
Wed Aug 24 13:22:13 MSD 2011


On 24/08/11 09:01, Sébastien Gay wrote:
> Hi,
>
> Thank you for your answers.
>
> @Alex
> The card is a Digium  TDM400P with 1 FXS and 3 FXO :
>
> active=yes
> alarms=OK
> description=Wildcard TDM400P REV I Board 5
> name=WCTDM/4
> manufacturer=Digium
> devicetype=Wildcard TDM400P REV I
> location=PCI Bus 00 Slot 07
> basechan=1
> totchans=4
> irq=17
> type=analog
> port=1,FXS
> port=2,FXO
> port=3,FXO
> port=4,FXO
>
> @ François
> I have tried some options in freetdm:
> <param name="answer-polarity-reverse" value="*true*" />
>
> <param name="hangup-polarity-reverse" value="*true*" />
>
> But the problem persists
>
>
> @Dario Garcia
>
> I did not know the option tone_detect, I'll do some tests.
>
>
> Thank you again for your help.
>
*
*Sébastien,
*
*Sounds like an issue I have been having. I wanted to have FS only act 
as a voicemail box, with the incoming land line connecting to both the 
FXO port and analog phones via a splitter. I wanted the voicemail to 
only pick up after 30s of ringing, however even if the incoming line had 
stopped ringing (and a polarity reverse was seen) the dialplan would 
still execute the VM and thus record a few seconds of dialtone. I saw 
the polarity reversal when the line stopped ringing but FS still claimed 
it was too close the the previous one even after more than 20s (the 
reversal time limit is set to something like 200ms) and did not hang up.

The VM would also kick in even when the analog phone was on a call after 
being picked up. There appears to be no way to test if an incoming 
analog line is still ringing or not (you can't do tone_detect here as 
the line is onhook).


Dialplan:

<include>
<extension name="public_ftdm">
<condition field="source" expression="freetdm">
<action application="sleep" data="30000"/>
<action application="answer"/>
<action application="voicemail" data="default ${domain_name} 1005"/>
</condition>
</extension>
</include>

However if I transfer the call to a SIP extension, remove the sleep and 
let the extension deal with the VM then it seems to work OK. Obviously 
this means my analog phones only ring for a moment though.

Alex

**

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