<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2722" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV><FONT face=Arial size=2>Hi again...</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> How are you
dialling this?</FONT></DIV>
<DIV><FONT face=Arial size=2>90446612345678 ? or 0446612345678 ?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> Also
another possibility is that the card is sending the DTMF when it haven't yet get
the tone from your PSTN? just thinking... in that case you can use the 'w' in
the dialstring to get a wait delay of 0.5 secs. (I don't think it might be this
one because your problem is speciffic to cells, but just thinking
aloud)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> Good
luck!</FONT></DIV>
<DIV><FONT face=Arial size=2>Alchaemist</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Claudio Canseco" <<A
href="mailto:claudio.canseco@gmail.com">claudio.canseco@gmail.com</A>>
wrote in message <A
href="news:8c1b1bde05092113458b6621@mail.gmail.com">news:8c1b1bde05092113458b6621@mail.gmail.com</A>...</DIV>
<DIV>Hi, thanks for your replay Alex:</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII
666MHz, 320 RAM) with no soundcard.</DIV>
<DIV>With an X100P clone card (an ambient modem).</DIV>
<DIV> </DIV>
<DIV>Everything looks good, I've been able to make local calls trough PSTN,
IAX, SIP.</DIV>
<DIV>I only have 1 POTS line, and 4 SIP softphones (X-lite) running all
right.</DIV>
<DIV>The only problem so far I have noticed (or realized of :P), it is
that i can make calls</DIV>
<DIV>to cellularphone numbers, * tries to connect but i get redirected to the
emergency service number 066.</DIV>
<DIV> </DIV>
<DIV>I don't think it is because of my dialplan, eventhough I tried several
configurations. Anyways here is part of the dialplan</DIV>
<DIV>where my softphones make calls:</DIV>
<DIV> </DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<P><BR>;########################################<BR>;# Llamadas
salientes [outgoing]
#<BR>;########################################</P>
<P>[outgoing]<BR>include => toPSTN<BR>include => iaxtel<BR>include
=> fwd-iax</P>
<P>;######## -> PSTN ########</P>
<P>[toPSTN]
; Permite hacer llamadas locales (7-digitos sin contar 9)<BR>ignorepat =>
9</P>
<P>exten => _92XXXXXX,1,NoOp("Call for "${EXTEN:1})<BR>exten =>
_92XXXXXX,2,Dial(Zap/1/${EXTEN:1})</P>
<P>exten => _904466XXXXXXXX,1,NoOp("Call for "(${EXTEN:1})
;Llamadas a Celular<BR>exten =>
_904466XXXXXXXX,2,Dial(Zap/1/ww${EXTEN:1})</P>
<P>;######## -> IAXTEL ########</P>
<P>[iaxtel]<BR>exten => _1700XXXXXXX,1,Dial(<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:IAX2/$%7Bcuenta%7D@iaxtel.com/$%7BEXTEN%7D@iaxtel"
target=_blank>IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel </A>)<BR>exten =>
_1888NXXXXXX,1,Dial(<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:IAX2/$%7Bcuenta%7D@iaxtel.com/$%7BEXTEN%7D@iaxtel"
target=_blank> IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel </A>)<BR>exten
=> _1877NXXXXXX,1,Dial(<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:IAX2/$%7Bcuenta%7D@iaxtel.com/$%7BEXTEN%7D@iaxtel"
target=_blank>IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel</A> )<BR>exten =>
_1866NXXXXXX,1,Dial( <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:IAX2/$%7Bcuenta%7D@iaxtel.com/$%7BEXTEN%7D@iaxtel"
target=_blank>IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel</A> )<BR>exten =>
_1800NXXXXXX,1,Dial(<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:IAX2/$%7Bcuenta%7D@iaxtel.com/$%7BEXTEN%7D@iaxtel"
target=_blank>IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel </A>)</P>
<P>;######## -> FWD ########</P>
<P>[fwd-iax]<BR>exten => _3.,1,SetCallerId,${FWDCIDNAME}<BR>exten =>
_3.,2,Dial(<A onclick="return top.js.OpenExtLink(window,event,this)"
href="http://iax2/$%7BFWDNUMBER%7D:$%7BFWDPASSWORD%7D@iax2.fwdnet.net/$%7BEXTEN:1%7D,60,tr)"
target=_blank>
IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60,tr)
</A><BR>exten => _3.,3,Congestion</P>
<P> </P>
<P>;#########################################<BR>;# Softphone
x-lite
#<BR>;#########################################</P>
<P>[x-lite] ; Note: SIP extensions are defined here as "66" followed
by any two digits<BR>include => default<BR>include =>
servicios<BR>include => outgoing</P>
<P>exten => 6600,1,NoOp(Llamada saliente maneja IAX2)<BR>exten =>
6600,2,Macro(dial,kano00,IAX2/kano00,20,tr)</P>
<P>exten => _X,1,NoOp(Llamada saliente maneja SIP)<BR>exten =>
_X,2,Macro(dial,667${EXTEN},SIP/667${EXTEN},20,tr)<BR></P></BLOCKQUOTE>
<P>All softphones working are SIP, and are directed to the [x-lite]
context.</P>
<P>This is my zapata.conf:</P>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<P>[channels]<BR>language=es<BR>context=incoming<BR>signalling=fxs_ks.<BR>usecallerid=yes<BR>hidecallerid=no<BR>callwaiting=yes<BR>callwaitingcallerid=yes<BR>threewaycalling=yes<BR>transfer=yes<BR>cancallforward=yes<BR>callreturn=yes
</P>
<P>echocancel=yes<BR>echocancelwhenbridged=yes<BR>echotraining=800<BR>rxgain=0.0<BR>txgain=25.0</P>
<P>group=1<BR>pickupgroup=1<BR>immediate=yes<BR>musiconhold=default<BR>relaxdtmf=yes
; Relajar el DTMF, poner si asterisk salta o duplica algún DTMF,
<BR> ; dando lugar a un número incorrecto.<BR>channel =>
1</P></BLOCKQUOTE>
<DIV dir=ltr>And my simple configuration file, </DIV>
<DIV dir=ltr>zaptel.conf:</DIV>
<DIV dir=ltr>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<P>loadzone=mx<BR>defaultzone=mx<BR>fxsks=1</P></BLOCKQUOTE>
<P>As you can see this aren't complicated configurations because i only have 1
X100P card, and I am currently using little extensions.<BR><BR>Also, I am not
using AMP but I'm thinking to installing it over my current
installation. I installed asterisk and zaptel from instructions i
got from several documentations sites (voip-info wiki, digium, etc).
<BR><BR>Well, I hope this info can help to look down the problem. Thanks
again,</P>
<P>Regards,</P>
<P>Claudio</P></DIV>
<P>
<HR>
<P></P>_______________________________________________<BR>--Bandwidth and
Colocation sponsored by Easynews.com --<BR><BR>Asterisk-Users mailing
list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To
UNSUBSCRIBE or update options visit:<BR>
http://lists.digium.com/mailman/listinfo/asterisk-users</BLOCKQUOTE></BODY></HTML>