[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
**
--
This message is intended only for the addressee and may contain
confidential information. Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.
"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/ebb9741d/attachment.html
Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users
mailing list