On Feb 12, 2008 10:40 AM, Ian <[EMAIL PROTECTED]> wrote: > > Hi all, > > its been quite a busy few day with pc's packing up etc, I recompile my > whole asterisk today using zaptel 1.4.7.1 and now the problem is > miraculously fixed, I will be sending this report to Digium bugs as well. > > Just a quick heads up for the order in which I had to recompile in order > for this to work > > > Recompile Zaptel > Restart Asterisk, asterisk doesn't pick up the zap channels > Recompile Libpri > Retart Asterisk, still no zap channels > Doing the thing I was hoping to skip, Recompile Asterisk > Everything in working order Did I miss something for me to have to only > recompile zaptel, or is that the way of doing things? > > Thank you all for your support > > Please scroll down to see the answers to my own stupid questions :-) >
Asterisk depends on Zaptel (well chan_zap and the respective codecs do) so always make sure to install first LibPRI, then Zaptel then Asterisk FWIW in the wav recording you sent there is alot of static. I am playing back with amaroK 1.4.7 of openSuSE. On Feb 12, 2008 11:50 AM, Andres Jimenez <[EMAIL PROTECTED]> wrote: > I am having similar problems running the same versions of Asterisk, > libpri & zaptel. > The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was > supossed to be related to FXO only, but I am having issues with a PRI > line and Digium's TE120P. > > Do you guys think it can be the same issue? > Maybe it is related but with PRI Asterisk does not generate any tone it sends a signal regarding your keypress. If you are using SIP phones make sure the dtmfmode in use is RFC2833. _______________________________________________ -- 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
