It has nothing to do with the GUI. Trixbox compiles Zaptel for you and provides them as RPMs for installation. Removing the RPMs and all the configs they leave lying around and compiling from source can be a complicated process, and the Trixbox forums/mailing lists will be better able to help the OP in this case.
<br><br>AR<br><br><div><span class="gmail_quote">On 5/15/07, <b class="gmail_sendername">Diego Iastrubni</b> <<a href="mailto:diego.iastrubni@xorcom.com">diego.iastrubni@xorcom.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On Tuesday 15 May 2007 19:11, Dave Cotton wrote:<br>> On Tue, 2007-05-15 at 17:45 +0200, Marco Vescovi wrote:<br>> > Hello,<br>> ><br>> > I'm writing because we have problems with an asterisk installation
<br>> > (Trixbox ver. 1.2.3). We have a customer which is receiving a lot o<br>> > telephony traffic (more or less 1 call/2 min.); we are using a TDM400<br>> > board, with 3 PSTN lines configured and we have two big issues:
<br>> ><br>> > - Calls are dropped during conversation (I have a busycount=8<br>> > from the initial value that was 4)<br>> ><br>> > - Sometimes when the user dials out, he hears the ringing tone
<br>> > but the line is already answered and the called party hears his voice<br>> > while he's still hearing the ringing tone.<br>> ><br>> > How can I investigate those 2 problems in order to find what's
<br>> > happening ?<br>><br>> Contact the Trixbox mailing lists?<br>Why is that? You think some fancy-shmancy GUI will fix this? The problem is<br>obviously in the zaptel area. But hey... this is "asterisk-users"...
<br><br>/me is in a fighting mode today<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>
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<br>Alex Robar<br><a href="mailto:alex.robar@gmail.com">alex.robar@gmail.com</a>