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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
