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On Fri, 2004-05-07 at 16:30, anthm@lists.digium.com wrote: > Update of /usr/cvsroot/asterisk/apps > In directory mongoose.digium.com:/tmp/cvs-serv17955/apps > added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call > +" 'D([digits])' -- Send DTMF digit string *after* called party has answered\n" > +" but before the brid...

I noticed the following came into cvs head yesterday: > Update of /usr/cvsroot/zaptel > In directory mongoose.digium.com:/tmp/cvs-serv2118 > > Modified Files: > fxotune.c wctdm.c wctdm.h > Log Message: > More TDM card echo API modifications. Making the fxotune program automatically > find the correct coefficients for the module. Lots of neat stuff. I did a complete cvs head checkout...

I've been getting these really weird errors that scroll almost nonstop across the screen when someone attempts to place a call and it starts ringing. Jul 19 17:55:01 ERROR[23246]: utils.c:509 tvfix: warning negative timestamp -536835.-70000 Does anyone know what this is all about and what I can do to fix it? Thanks for any help in advance, this is causing huge problems!

...e: [Asterisk-cvs] asterisk utils.c,1.59,1.60 > c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] > > ---------------------------------------------------------------------- > ------ > ---- > > Update of /usr/cvsroot/asterisk/include/asterisk > In directory mongoose.digium.com:/tmp/cvs-serv17202/include/asterisk > > Modified Files: > time.h > Log Message: > fix negative timestamp issue > > ......http://lists.digium.com/pipermail/asterisk-cvs/2005-July/ > 007087.html Rock on. Thanks Russell!

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

Hi, I want to use the cdr to record the call log to my Microsoft SQL Server using unixodbc and freetds.... but when I compile, I've got this message.... Does anyone have the same problem and/or know how to solve it ? Thanks Baste regards David Masure make[1]: Entering directory `/usr/src/asterisk/bristuff-0.2.0-RC7k/asterisk-1.0.6/cdr' gcc -pipe -Wall -Wstrict-prototypes

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

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...ath: <ho@routefree.com> Delivered-To: samba@lists.samba.org Received: from therouter.routefree.com (unknown [209.21.47.229]) by lists.samba.org (Postfix) with ESMTP id 631214128 for <samba@lists.samba.org>; Thu, 6 Sep 2001 16:22:23 -0700 (PDT) Received: from routefree.com (IDENT:ho@mongoose.routefree.com [10.0.0.201]) by therouter.routefree.com (8.9.3/8.9.3) with ESMTP id QAA26855; Thu, 6 Sep 2001 16:26:31 -0700 Message-ID: <3B980627.277BA2FB@routefree.com> From: "W. Wilson Ho" <ho@routefree.com> Organization: RouteFree, Inc. X-Mailer: Mozilla 4.76 [en] (X11;...