Hi Dave, that actually makes sense.. I had probs in figuring out my disconnection dial tone, till the point I stoped trying to figure out.. so ur right that might be the problem.. thanks for your help ill give it a try :)
best, Roland -------------------------------------------------- From: "Dave Fullerton" <[EMAIL PROTECTED]> Sent: Wednesday, November 12, 2008 8:23 PM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: Re: [asterisk-users] PSTN Channels merging with SIP channels!!! > [EMAIL PROTECTED] wrote: >> Hi All, >> >> I appreciate any help with this issue am facing. >> >> first of all my topology is as such: >> >> my asterisk box has two callcentric sip accounts on it. >> as well as a PSTN line which is connected to asterisk through a Sipura >> 3102. >> >> now my problem is as such: >> >> I sometimes use my box as an international gateway. >> that means when am home, I call my PSTN line. it directs me through an >> auto attendant that I've setup. >> in it there's a "waitexten" for 8 seconds, where I enter the full >> international number with a preceding extension which directs this number >> to one of my callcentric lines. >> >> now this has worked ok for a while but lately am facing a major prob! >> sometimes ill be talking to an international destination following the >> scenario explained above. and suddenly In the middle of my conversation >> both me and the other side could hear my PSTN attendant's recorded voice >> welcoming caller.. >> such a thing is extremely annoying to say the least and I have no idea >> why it's happening.. >> can anyone help out please?! any advice on how to catch what's causing >> this?! > > I had something similar happen to me using a SPA 3000 that I was using > to bridge two PBX's together. I don't remember exactly how I resolved > it, but the issue had something to do with the SPA not correctly > reporting that it was busy. For example, if someone rang the PSTN port > on the SPA it would answer and connect to an auto attendant. If asterisk > then tried to make a call through said PSTN port rather than the SPA > saying it was busy sometimes it would bridge the calls together and play > the auto attendant message like you're describing. I think I ended up > using CHANISAVAIL() in my dialplan to see if the SPA was busy rather > than relying on the result of Dial(). > > Hopefully that's of some help to you. > > -Dave > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
