[Asterisk-Users] A Major Problem!

[Steven Critchfield]

On Fri, 2003-05-30 at 08:23, Surajee Ratnayake wrote:
> yes, that solves the problem, thank you very much,
> but my other problem remains, will this be a problem when it comes to E1
> lines?
> i am very sorry for keep on asking this

Depends on the signalling of your E1 line. If you are doing a PRI(PRA?)
you will have absolute signaling that will bring up and down phone lines
with no problems. If you are using a RBS type line, it will depend on
the type of signalling they are providing to you. 

> ----- Original Message -----
> From: "Surajee Ratnayake" <surajee at infotechs.lk>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, May 30, 2003 7:04 PM
> Subject: Re: [Asterisk-Users] A Major Problem!
> 
> 
> > no, we dont have a "busydetect=yes" line in the zapata.conf, we will put
> it
> > and giv it a try,
> > btw, what will be the case with an E1 line, will the same problem occur?
> >
> > Surajee
> >
> > ----- Original Message -----
> > From: "Michael Bielicki" <Michael.Bielicki at Global-Gateway.net>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Friday, May 30, 2003 6:51 PM
> > Subject: Re: [Asterisk-Users] A Major Problem!
> >
> >
> > > Do you have busydetect set to yes in zapata,.comf ? uou need that for
> > analog
> > > lines and you cannot have that for E1 lines :)
> > >
> > > regards
> > >
> > > Michael Bielicki
> > >
> > > On Friday 30 May 2003 1:38 pm, Surajee Ratnayake wrote:
> > > > hi,
> > > >
> > > > we are experiecing the following probem, if anybody have come across
> > such a
> > > > problem or a solution to this please let us know. our set up is, an
> > > > Asterisk server equipped with, 4 port station interface card ,single
> > port
> > > > fxo card and several soft sip phones we have found problems with the
> > > > following scenarios,
> > > >
> > > > outside caller (calling through fxo interface)
> > > > <------------------------------>  sip phone/ station interface phone
> > > >
> > > >
> > calls
> > > > to a conference outside caller (calling through fxo
> > > > interface)<--------------------------------------------------->
> > confernce
> > > >
> > > > the problem is, once the outside caller(calling through fxo interface)
> > > > disconnects the line, Asterisk does not detects the disconnection,
> other
> > > > party can hear the 'engage like tone' coming from the other side.This
> > > > continues till the other party(probalby the sip phone or the station
> > > > interface phone) hangs up. If the fxo user was in a conference if he
> > > > disconnets the line, other confencees can here the 'engage like tone'
> ,
> > > > this is very disturbing. The biggest problem is, the fxo line remains
> > busy,
> > > > till the sip/station phone user disconnects the line. Can anybody give
> > us a
> > > > solution for this.
> > > >
> > > > In the near future, we are going to add some E1 lines too(with E400P
> > > > cards), once this is done, will the above call disconnection problem
> > occur
> > > > in that configuration too..or is this a common problem only with
> analog
> > ?
> > > >
> > > > Thank you very much,
> > > > Surajee
> > >
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> >
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> >
> >
> 
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-- 
Steven Critchfield  <critch at basesys.com>