[asterisk-users] Asterisk, POTS and plain handsets

[Steve Totaro]

Ah, I am not sure why what you want to accomplish cannot be done or
why it shouldn't be.  Like you said, it worked with wctdm.

When I tried it you had to modprobe wcfxo and wcfxs, back in the ole
days, and it was confusing which to load first........  The behavior
was the same as you describe with your new card.  I never tried it
with wctdm.

Thanks,
Steve T

On Fri, Jun 27, 2008 at 4:59 PM, Steve <astuser at braingia.org> wrote:
>
> Upon further digging, this is seems almost certainly related to the card
> and kernel module being used for the card.  I loaded up the old TDM22B
> card and I'm no longer having the issue.  I even took one of the FXO
> (red) modules from the new card and put it into the old card and it
> still works.
>
> When the new card loads it uses kernel module wctdm24xxp for its kernel
> module whereas the old card used wctdm for its module.  So there must be
> some flaw in the wctdm24xxp module -- or some flaw in wctdm was
> fixed but it was really a _feature_ to me.  :)
>
> I'm wondering if this isn't worth a bug report though -- or maybe
> there's something wrong with the new card?
>
> Steve
>
> On Fri, Jun 27, 2008 at 08:49:48AM -0500, Steve wrote:
>> On Thu, Jun 26, 2008 at 10:35:13PM -0400, Steve Totaro wrote:
>> > Post the output from Asterisk's CLI.  I think maybe your contexts are
>> > overlapping or are the same.  It should say something to the effect of
>> > "Starting simple switch"
>>
>> When I take one of the plain phones off-hook, just lifting the
>> receiver, here's the CLI output.  Zap3/1 is an FXO port which is
>> connected to the POTS line.  Note that sometimes taking the phone
>> off-hook doesn't do it but only when the receiver is hung up or put back
>> on-hook does asterisk start to detect a ring.
>>
>> The context home-incoming has one thing to do, dial a SIP phone for 20
>> seconds.  On normal incoming calls this works great.
>>
>>     -- Starting simple switch on 'Zap/3-1'
>> [Jun 27 08:31:16] ERROR[8889]: callerid.c:540 callerid_feed: No start
>> bit found in fsk data.
>>     -- Executing [s at home-incoming:1] Dial("Zap/3-1", "SIP/gxp20001,20")
>> in new stack
>>   == Using SIP RTP CoS mark 5
>>     -- Called gxp20001
>>     -- SIP/gxp20001-081d50c0 is ringing
>>   == Spawn extension (home-incoming, s, 1) exited non-zero on 'Zap/3-1'
>>     -- Hungup 'Zap/3-1'
>>
>> > Check what context your FXO channels are in, something like
>> > context=from-verizon and then check the context of your FXS (plain
>> > telephones), they should be in a different context such as
>> > context=to-phones.
>>
>> The FXO ports come from the telco drop to the asterisk server.  The card
>> doesn't have any FXS ports on it rather these phones are also connected
>> directly to the telco drop.  So /something/ is happening where when one
>> of these phones is taken off-hook (or sometimes put back on-hook),
>> asterisk is catching it as being interesting event.
>>
>> > Also, make sure you have immediate=no
>>
>> Didn't have it directly in the channel config, added it and rebooted and
>> no luck.
>>
>> > Then check your dialplan and make sure those contexts do what you want
>> > and you are not accidentally including a context where it should not
>> > be.
>>
>> The contexts themselves do what I want but for some reason asterisk is
>> trying to handle an off-hook situation anywhere else on the copper as an
>> incoming call and handling it through the home-incoming context.
>>
>> My hope is to be able to have asterisk act as an answering machine would
>> on this line, sharing the line with other regular phones.  When a phone
>> call comes in, I'd like it to ring the two SIP phones on the network and
>> then if no one picks up after N seconds, answer the call.  Obviously,
>> the regular phones would ring normally on an incoming call because
>> they're still connected to the telco drop.
>>
>> Thanks for any further assistance.
>>
>> Steve
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>