Thanks for the response. Answers inline.. -----Original Message----- >From: Michael Neuhauser <[EMAIL PROTECTED]> >Sent: Oct 3, 2006 10:37 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion ><[email protected]>, Clif Jones <[EMAIL PROTECTED]> >Subject: Re: [asterisk-users] Problems with automon > >On Tue, 2006-10-03 at 09:20 -0400, Clif Jones wrote: >> I have been trying to get automon working on Asterisk 1.2.12.1 and I >> am having some problems. I have searched the list archives and have >> not found my answer either. This system is setup for SIP to SIP calls >> with G.729 codecs. > >You do have a G.729 codec module on your machine? Yes, I have plenty of G.729 licenses and of course, the commercial G.729 codec. > >> I believe that I have the config files setup (*1 enabled in >> features.conf, DYNAMIC_FEATURES global variable and 'wW' Dial options >> in extensions.conf). > >Are you using [featuremap]/automon or [applicationmap]/<MYOWNFEATURE>? >(in the first case you do not need DYNAMIC_FEATURES in the second case >the wW flags do apply). I'm using the [featuremap] with the example code for '*1 - automon' that was provided. So, you are saying that I do NOT need the DYNAMIC_FEATURES global OR the 'wW' options for the [featuremap]? I'm trying to think back but I believe that I had trouble getting this working without some of that but I may be wrong. > >> I lose audio both ways when '*1' is pressed and if I hangup on the >> originating side that initiated automon, I can hear a brief DTMF tone >> on the gateway side just before the call is dropped. I also get the >> following log: >> Sep 17 21:17:21 WARNING[3007] res_features.c: Bridge failed on >> channels SIP/2009-081eb5b8 and SIP/gateway-081e58c0 > >This could be related to a bug in 1.2.12.1 that affects DTMF handling >for bridged calls (see http://bugs.digium.com/view.php?id=7982). Bug was >fixed in 1.2 branch -> get 1.2 branch from SVN or wait for the 1.2.13 >release. Automon does work for me after the fix from the 1.2 branch was >applied. I'll look into this. Thanks! I'm still trying to figure out though why it works for some cases and not others. I'm using RFC2833 DTMF in all cases.
>-- >Dr. Michael Neuhauser mailto:[EMAIL PROTECTED] >Firmix Software GmbH sip:[EMAIL PROTECTED] >Vienna/Austria/Europe tel:+43-1-7890849-30 >Linux Development and Services http://www.firmix.at/ > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
