<html><div style='background-color:'><P>Not really sure, but once I had a problem when I changed the txgain and rxgain, so set them again to 0.0 and see how it will work.<BR><BR></P>
<P>Truely/</P>
<P>Ammar</P>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #a0c6e5 2px solid; MARGIN-RIGHT: 0px"><FONT style="FONT-SIZE: 11px; FONT-FAMILY: tahoma,sans-serif">
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From: <I>"Jerome SOUCANY" <soucany@app-line.com></I><BR>Reply-To: <I>Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com></I><BR>To: <I><asterisk-users@lists.digium.com></I><BR>Subject: <I>[Asterisk-Users] No sound on 10% of incoming calls</I><BR>Date: <I>Tue, 7 Feb 2006 11:03:49 +0100</I><BR>>Hello,<BR>><BR>>I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring<BR>>but I don't hear the caller and the caller doesn't hear me (all IP Phones<BR>>have the same problem).<BR>><BR>>This problem appear also if the call is directly send to the second E1 of<BR>>the digium card who is connected to an IVR.<BR>><BR>>It does not depand on the charge of the server (I have the problem with only<BR>>one call).<BR>><BR>>The configuration :<BR>><BR>>PRI (France Telecom) 15 channels
<====> Asterisk <=====> IP Phone<BR>><BR>>* Server :<BR>> - Dell power edge 1800SC<BR>> - 2 Ethernet cards (LAN + VoIP LAN)<BR>> - Digium card : TE 405P<BR>> - Linux Mandriva LE 2005 (10.2) :<BR>> Linux ASTERISK 2.6.11-12mdksmp #1 SMP i686 Intel(R) Xeon(TM) CPU<BR>>3.00GHz unknown GNU/Linux<BR>> - Asterisk 1.2.4<BR>> - Zaptel 1.2.3<BR>> - Libpri 1.2.2<BR>><BR>>* IP Phone :<BR>> SNOM 320 (latest firmware)<BR>><BR>>============================================<BR>>zaptel.conf<BR>><BR>>span=1,1,0,ccs,hdb3<BR>>span=2,1,0,ccs,hdb3,crc4,yellow<BR>>span=3,1,0,ccs,hdb3,crc4,yellow<BR>>span=4,1,0,ccs,hdb3,crc4,yellow<BR>><BR>>bchan = 1-15, 17-31<BR>>dchan = 16<BR>>bchan = 32-46,48-62<BR>>dchan = 47<BR>>bchan = 63-77,79-93<BR>>dchan = 78<BR>>bchan = 94-108,110-124<BR>>dchan =
109<BR>><BR>>loadzone = fr<BR>>defaultzone = fr<BR>><BR>>============================================<BR>><BR>>============================================<BR>>zapata.conf<BR>><BR>>[channels]<BR>>switchtype=euroisdn<BR>>pridialplan=national<BR>>signalling=pri_cpe<BR>>usecallerid=yes<BR>>hidecallerid=yes<BR>>usecallingpres=no<BR>>callwaiting=yes<BR>>callwaitingcallerid=yes<BR>>threewaycalling=yes<BR>>transfer=yes<BR>>cancallforward=yes<BR>>echocancel=yes<BR>>echocancelwhenbridged=yes<BR>>echotraining=yes<BR>>rxgain=0.0<BR>>txgain=-6.0<BR>><BR>>group=1<BR>>callgroup=1<BR>>pickupgroup=1<BR>><BR>>immediate=no<BR>>callprogress=yes<BR>><BR>>callerid=asreceived<BR>>group=1<BR>>context=from-pstn<BR>>signalling=pri_cpe<BR>>channel => 1-15 ;,17-31 => only 15 first channels on
PRI<BR>><BR>>group=2<BR>>context=from-ivr<BR>>signalling=pri_net<BR>>channel => 32-46,48-62<BR>><BR>>group=3<BR>>context=from-ivr-bis<BR>>signalling=pri_net<BR>>channel => 63-77,79-93<BR>><BR>>group=4<BR>>signalling=pri_net<BR>>channel => 94-108,110-124<BR>>============================================<BR>><BR>><BR>><BR>><BR>>Any ideas ?<BR>><BR>><BR>><BR>>Regards<BR>><BR>>Jerome<BR>><BR>><BR>>_______________________________________________<BR>>--Bandwidth and Colocation provided by Easynews.com --<BR>><BR>>Asterisk-Users mailing list<BR>>To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR></FONT></BLOCKQUOTE></div><br clear=all><hr>Open your e-mail without having to worry about viruses with <a href="http://g.msn.com/8HMAENCA/2737??PS=47575" target="_top">MSN Premium.</a> Join now and get the first two months FREE*</html>