[Asterisk-Users] TDM400P digium card

[Dewey Straughn]

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?
 
-Dewey
 

 
________________________________

From: asterisk-users-bounces at lists.digium.com on behalf of Nora Lavelle
Sent: Mon 2/27/2006 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TDM400P digium card 



Hi Dewey - 

 

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 ? ;-) 

 

zapata.conf:

;

; Zapata telephony interface

;

; Configuration file

 

[channels]

language=en

context=default

switchtype=national

signalling=fxs_ks

usecallerid=yes

hidecallerid=no

callwaiting=yes

callwaitingcallerid=yes

threewaycalling=yes

transfer=yes

cancallforward=yes

callreturn=yes

echocancel=yes

echocancelwhenbridged=yes

echotraining=800

rxgain=0.0

txgain=0.0

group=1

immediate=yes

channel => 1,2,3,4

 

extensions.conf:

 

[incoming]

exten => s,1,Answer();

exten => s,2,Background(ssn-greeting);

exten => *,1,Directory(default)

exten => 205,1,Wait(2)

exten => 205,2,Record(/tmp/asterisk-recording:gsm)

exten => 205,3,Wait(2)

exten => 205,4,Playback(/tmp/asterisk-recording)

exten => 205,5,Wait(2)

exten => 205,6,Hangup 

 

[internal]

exten => 101,1,Macro(stdexten,SIP/101)

exten => 102,1,Macro(stdexten,SIP/102)

exten => 103,1,Macro(stdexten,SIP/103)

exten => 123,1,Macro(stdexten,SIP/123)

exten => 124,1,Macro(stdexten,SIP/124)

exten => 125,1,Macro(stdexten,SIP/125)

exten => 126,1,Macro(stdexten,SIP/126)

exten => 127,1,Macro(stdexten,SIP/127)

exten => 128,1,Macro(stdexten,SIP/128)

exten => 129,1,Macro(stdexten,SIP/129)

exten => 130,1,Macro(stdexten,SIP/130)

exten => 135,1,Macro(stdexten,SIP/135)

exten => 117,1,Macro(stdexten,SIP/117)

exten => 201,1,Macro(stdexten,SIP/201)

 

; Please begin new extensions here

exten => 250,1,Macro(stdexten,SIP/250)

 

[voicemail]

exten => 300,1,Ringing

exten => 300,2,Wait(2)

exten => 300,3,System(/var/spool/asterisk/vm/fix_volume.pl)

exten => 300,4,VoicemailMain(ssn-voicemail-greeting)

exten => 300,5,Hangup

 

[local]

exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})

exten => _9NXXXXXX,2,Congestion

 

[longdistance]

exten => _91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})

exten => _91NXXNXXXXXX,2,Congestion

 

; exten => s,103,Hangup

 

[macro-stdexten]

exten => s,1,Dial(${ARG1},20)

exten => s,2,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})

exten => s-NOANSWER,2,Goto(default,s,1)

exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})

exten => s-BUSY,2,Goto(default,s,1)

exten => s-CONGESTION,1,Voicemail(b${MACRO_EXTEN})

exten => s-CONGESTION,2,Goto(default,s,1)

exten => s-.,1,Goto(s-NOANSWER,1)

exten => a,1,VoicemailMain(${MACRO_EXTEN})

 

[default]

include => incoming

include => internal

include => voicemail

include => local

include => longdistance

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