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<title>RE: [Asterisk-Users] TDM400P digium card </title>
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><!-- Converted from text/plain format -->I’d
suggest testing each set using a virtual “echo test” extension, this
will test your network between sets and asterisk box. If you have choppy
audio during the test, then I would look into the LAN for issues.</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>-vince</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Nora Lavelle<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, February 27, 2006
5:15 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial
Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Asterisk-Users]
TDM400P digium card </span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> </span></font></p>
<p style='margin-right:0in;margin-bottom:12.0pt;margin-left:.5in'><font size=2
face="Times New Roman"><span style='font-size:10.0pt'>Thank you so much. we
aren't using wireless. just our LAN. So it sounds like network connectivity and
possibly my asterisk server. Thanks for helping to pointme in the right
direction I so apprciate it.<br>
<br>
-nora<br>
<br>
<br>
-----Original Message-----<br>
From: asterisk-users-bounces@lists.digium.com on behalf of Dewey Straughn<br>
Sent: Mon 2/27/2006 4:20 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: RE: [Asterisk-Users] TDM400P digium card<br>
<br>
<br>
Unless I am missing something, it looks like you only use pots (Plain Old
Telephone Service) lines for making and receiving calls correct? It doesn't
appear you are using and VOIP termination (Making outbound calls) or
origination (Receiving inbound calls) provider. If you are getting choppy calls
and your extensions are not outside your LAN, you need to troubleshoot you lan
and Asterisk server. Make sure you can ping it with no packet loss or high
latency. That's were I would start. Using a basic configuration (IE. POTS
lines, TDM400, all lan extensions), you really shouldn't have any issues to
deal with. It's pretty straight forward. Is any of this wirelessly connected?<br>
<br>
-Dewey<br>
<br>
<br>
<br>
_____ <br>
<br>
From: asterisk-users-bounces@lists.digium.com on behalf of Nora Lavelle<br>
Sent: Mon 2/27/2006 5:48 PM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: RE: [Asterisk-Users] TDM400P digium card<br>
<br>
<br>
<br>
Hi Dewey -<br>
<br>
<br>
<br>
So as those who read this list know I'm very new to voip software. So as
embarrassed as I am to say it. I don't know how to answer all of your questions.
I have no idea how many voip trunks I have or if I'm using G.729. We have
a DSL connection currently. I have 4 analog phone lines connected to a
digium card that's plugged into a dell Here's my Zapata.conf and
extensions.conf file. I'm definitely confused here. Can y'all tell ? ;-)<br>
<br>
<br>
<br>
zapata.conf:<br>
<br>
;<br>
<br>
; Zapata telephony interface<br>
<br>
;<br>
<br>
; Configuration file<br>
<br>
<br>
<br>
[channels]<br>
<br>
language=en<br>
<br>
context=default<br>
<br>
switchtype=national<br>
<br>
signalling=fxs_ks<br>
<br>
usecallerid=yes<br>
<br>
hidecallerid=no<br>
<br>
callwaiting=yes<br>
<br>
callwaitingcallerid=yes<br>
<br>
threewaycalling=yes<br>
<br>
transfer=yes<br>
<br>
cancallforward=yes<br>
<br>
callreturn=yes<br>
<br>
echocancel=yes<br>
<br>
echocancelwhenbridged=yes<br>
<br>
echotraining=800<br>
<br>
rxgain=0.0<br>
<br>
txgain=0.0<br>
<br>
group=1<br>
<br>
immediate=yes<br>
<br>
channel => 1,2,3,4<br>
<br>
<br>
<br>
extensions.conf:<br>
<br>
<br>
<br>
[incoming]<br>
<br>
exten => s,1,Answer();<br>
<br>
exten => s,2,Background(ssn-greeting);<br>
<br>
exten => *,1,Directory(default)<br>
<br>
exten => 205,1,Wait(2)<br>
<br>
exten => 205,2,Record(/tmp/asterisk-recording:gsm)<br>
<br>
exten => 205,3,Wait(2)<br>
<br>
exten => 205,4,Playback(/tmp/asterisk-recording)<br>
<br>
exten => 205,5,Wait(2)<br>
<br>
exten => 205,6,Hangup<br>
<br>
<br>
<br>
[internal]<br>
<br>
exten => 101,1,Macro(stdexten,SIP/101)<br>
<br>
exten => 102,1,Macro(stdexten,SIP/102)<br>
<br>
exten => 103,1,Macro(stdexten,SIP/103)<br>
<br>
exten => 123,1,Macro(stdexten,SIP/123)<br>
<br>
exten => 124,1,Macro(stdexten,SIP/124)<br>
<br>
exten => 125,1,Macro(stdexten,SIP/125)<br>
<br>
exten => 126,1,Macro(stdexten,SIP/126)<br>
<br>
exten => 127,1,Macro(stdexten,SIP/127)<br>
<br>
exten => 128,1,Macro(stdexten,SIP/128)<br>
<br>
exten => 129,1,Macro(stdexten,SIP/129)<br>
<br>
exten => 130,1,Macro(stdexten,SIP/130)<br>
<br>
exten => 135,1,Macro(stdexten,SIP/135)<br>
<br>
exten => 117,1,Macro(stdexten,SIP/117)<br>
<br>
exten => 201,1,Macro(stdexten,SIP/201)<br>
<br>
<br>
<br>
; Please begin new extensions here<br>
<br>
exten => 250,1,Macro(stdexten,SIP/250)<br>
<br>
<br>
<br>
[voicemail]<br>
<br>
exten => 300,1,Ringing<br>
<br>
exten => 300,2,Wait(2)<br>
<br>
exten => 300,3,System(/var/spool/asterisk/vm/fix_volume.pl)<br>
<br>
exten => 300,4,VoicemailMain(ssn-voicemail-greeting)<br>
<br>
exten => 300,5,Hangup<br>
<br>
<br>
<br>
[local]<br>
<br>
exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})<br>
<br>
exten => _9NXXXXXX,2,Congestion<br>
<br>
<br>
<br>
[longdistance]<br>
<br>
exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})<br>
<br>
exten => _91NXXNXXXXXX,2,Congestion<br>
<br>
<br>
<br>
; exten => s,103,Hangup<br>
<br>
<br>
<br>
[macro-stdexten]<br>
<br>
exten => s,1,Dial(${ARG1},20)<br>
<br>
exten => s,2,Goto(s-${DIALSTATUS},1)<br>
<br>
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})<br>
<br>
exten => s-NOANSWER,2,Goto(default,s,1)<br>
<br>
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})<br>
<br>
exten => s-BUSY,2,Goto(default,s,1)<br>
<br>
exten => s-CONGESTION,1,Voicemail(b${MACRO_EXTEN})<br>
<br>
exten => s-CONGESTION,2,Goto(default,s,1)<br>
<br>
exten => s-.,1,Goto(s-NOANSWER,1)<br>
<br>
exten => a,1,VoicemailMain(${MACRO_EXTEN})<br>
<br>
<br>
<br>
[default]<br>
<br>
include => incoming<br>
<br>
include => internal<br>
<br>
include => voicemail<br>
<br>
include => local<br>
<br>
include => longdistance<br>
<br>
<br>
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