Hi Vinicius,
on this link you have an explanation about your problem.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Basically, you need to know what method your pbx is using for signaling the end
of the call, and then, you must to setup the zaptel driver and zapata channel
according to that.
chers
Fabay
-----Mensaje original-----
De: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]En nombre de Vin?cius
Fontes - CANALL
Enviado el: Mi?rcoles, 17 de Mayo de 2006 08:55 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] TDM does not disconnect
Hello all.
This is my very first message to the list. I have a TDM400P card, It
has 2 FXO channels which are connected to extensions of my PBX
(Ericsson BP250), so I can dial from any SIP softphone directly to
physical (analog and digital) extensions on my company.
My PBX is configured so when I dial 8 on any extension, it will
redirect to the first free FXO channel on my TDM400P card. Then I use
the Asterisk's DISA application to get a dial tone, like this:
exten => s,1,disa(no-password,tdm-disa)
[tdm-disa]
exten => _XXX.,1,ChanIsAvail(Zap/3&Zap/4) ; Checks for a free channel to
dial
exten => _XXX.,2,Dial(${AVAILORIGCHAN}/${EXTEN}) ; Dials the number on
the first channel available
But if the person I'm calling does not answer the phone and I hangup
(fisically) the extension, the Zap channels doesn't hangup! They stay
connected, and the line I called keeps on ringing.
So, this is the entire process:
1. I pickup a physical extension, and dial 8
2. The PBX redirects the call to the first FXO channel available
3. Asterisk answers the call and gives a dial tone using the DISA application
4. I dial the number I want
5. Asterisk dials using an available Zap channel
6. If the person I called does not answer the phone, I hangup my
extension but the FXO channels doesn't hangup!
This is the logs I got running asterisk -vvvvvvvvvvvvvvv on the
situation above. My comments on it are rounded with []:
[I pickup my physical extension and dial 8]
-- Starting simple switch on 'Zap/3-1'
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18
(Ring Begin)...
May 17 08:48:52 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 2
(Ring/Answered)...
May 17 08:48:56 NOTICE[16335]: chan_zap.c:6184 ss_thread: Got event 18
(Ring Begin)...
-- Executing DISA("Zap/3-1", "no-password|tdm-disa") in
new stack
[Asterisk gives me dial tone and I dial 081168345 -> 0 + my cell phone
number]
-- Executing ChanIsAvail("Zap/3-1", "Zap/3&Zap/4")
in new stack
-- Hungup 'Zap/4-1'
-- Executing NoOp("Zap/3-1", "Canal: Zap/4") in new
stack
-- Executing Dial("Zap/3-1", "Zap/4/081168345") in new
stack
-- Called 4/081168345
[My cell phone starts to ring, I hangup my extension. Cell phone keeps
on ringing.]
[After a while (about one minute) the following shows up]
-- Zap/4-1 is busy
-- Hungup 'Zap/4-1'
== Everyone is busy/congested at this time (1:1/0/0)
-- Hungup 'Zap/3-1'
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users