<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.10.0">
</HEAD>
<BODY>
Once I got the wires plugged into the right ports, I discovered ztcfg wasn't running, fixed that. Now a softphone can call out but a handset can't.<BR>
<BR>
Here is a little more info:<BR>
# asterisk -vvvvvvr<BR>
Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.<BR>
Created by Mark Spencer <<A HREF="mailto:markster@digium.com">markster@digium.com</A>><BR>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.<BR>
This is free software, with components licensed under the GNU General Public<BR>
License version 2 and other licenses; you are welcome to redistribute it under<BR>
certain conditions. Type 'core show license' for details.<BR>
=========================================================================<BR>
== Parsing '/etc/asterisk/asterisk.conf': Found<BR>
== Parsing '/etc/asterisk/extconfig.conf': Found<BR>
Connected to Asterisk 1.4.21.2 currently running on asterisk (pid = 3223)<BR>
Verbosity is at least 6<BR>
<BR>
asterisk*CLI> zap show channels<BR>
Chan Extension Context Language MOH Interpret<BR>
pseudo DLPN_DialPlan1 default<BR>
1 DID_trunk_1 default<BR>
4 DLPN_DialPlan1 default<BR>
<BR>
With the handset on hook:<BR>
asterisk*CLI> zap show channel 4<BR>
Channel: 4LI><BR>
File Descriptor: 25<BR>
Span: 1k*CLI><BR>
Extension: I><BR>
Dialing: noI><BR>
Context: DLPN_DialPlan1<BR>
Caller ID: 6002<BR>
Calling TON: 0<BR>
Caller ID name: Bob Crandell<BR>
Destroy: 0LI><BR>
InAlarm: 0LI><BR>
Signalling Type: FXO Kewlstart<BR>
Radio: 0*CLI><BR>
Owner: <None><BR>
Real: <None>><BR>
Callwait: <None><BR>
Threeway: <None><BR>
Confno: -1LI><BR>
Propagated Conference: -1<BR>
Real in conference: 0<BR>
DSP: nok*CLI><BR>
Relax DTMF: no<BR>
Dialing/CallwaitCAS: 0/0<BR>
Default law: ulaw<BR>
Fax Handled: no<BR>
Pulse phone: no<BR>
Echo Cancellation: 128 taps, currently OFF<BR>
Actual Confinfo: Num/0, Mode/0x0000<BR>
Actual Confmute: No<BR>
Hookstate (FXS only): Onhook<BR>
<BR>
With the handset off hook:<BR>
asterisk*CLI> zap show channel 4<BR>
Channel: 4LI><BR>
File Descriptor: 25<BR>
Span: 1k*CLI><BR>
Extension: I><BR>
Dialing: noI><BR>
Context: DLPN_DialPlan1<BR>
Caller ID: 6002<BR>
Calling TON: 0<BR>
Caller ID name: Bob Crandell<BR>
Destroy: 0LI><BR>
InAlarm: 0LI><BR>
Signalling Type: FXO Kewlstart<BR>
Radio: 0*CLI><BR>
Owner: <None><BR>
Real: <None>><BR>
Callwait: <None><BR>
Threeway: <None><BR>
Confno: -1LI><BR>
Propagated Conference: -1<BR>
Real in conference: 0<BR>
DSP: nok*CLI><BR>
Relax DTMF: no<BR>
Dialing/CallwaitCAS: 0/0<BR>
Default law: ulaw<BR>
Fax Handled: no<BR>
Pulse phone: no<BR>
Echo Cancellation: 128 taps, currently OFF<BR>
Actual Confinfo: Num/0, Mode/0x0000<BR>
Actual Confmute: No<BR>
Hookstate (FXS only): Onhook<BR>
<BR>
<BR>
On Tue, 2009-02-17 at 14:05 -0600, Ryan Brindley wrote:
<BLOCKQUOTE TYPE=CITE>
<FONT COLOR="#000000">Bob,</FONT><BR>
<FONT COLOR="#000000">If you're in Asterisk's CLI, make sure you have verbosity past 3 ('core set verbose 3') and then try to make an outbound call. You should see a bit of dialplan scroll on the screen and look for errors there. Copy and paste em here if you find any and don't know what to do.</FONT><BR>
<FONT COLOR="#000000">-- </FONT><BR>
<FONT COLOR="#000000">Ryan Brindley </FONT><BR>
<FONT COLOR="#000000">Digium, Inc. | Software Developer </FONT><BR>
<FONT COLOR="#000000">445 Jan Davis Drive NW - Huntsville, AL 35806 - USA </FONT><BR>
<FONT COLOR="#000000">main: +1 256-428-6000 fax: +1 256-864-0464 </FONT><BR>
<FONT COLOR="#000000">Check us out at: http://digium.com & http://asterisk.org</FONT><BR>
<BR>
<FONT COLOR="#000000">----- Original Message -----</FONT><BR>
<FONT COLOR="#000000">From: "Bob Crandell" <bob@assuredcomp.com></FONT><BR>
<FONT COLOR="#000000">To: "Asterisk GUI" <asterisk-gui@lists.digium.com></FONT><BR>
<FONT COLOR="#000000">Sent: Tuesday, February 17, 2009 1:30:07 PM GMT -06:00 US/Canada Central</FONT><BR>
<FONT COLOR="#000000">Subject: [asterisk-gui] Dialing out</FONT><BR>
<BR>
<FONT COLOR="#000000">Hi All,</FONT><BR>
<BR>
<FONT COLOR="#000000">Fresh install. There are 2 Digium cards, 1 with 2 FXO ports and 1 with</FONT><BR>
<FONT COLOR="#000000">2 FXS ports. These show up in "Configure Hardware". I added a trunk</FONT><BR>
<FONT COLOR="#000000">with these ports, configured Outgoing Calling Rules, setup a dial plan</FONT><BR>
<FONT COLOR="#000000">that included everything and added 2 users which can call each other.</FONT><BR>
<BR>
<FONT COLOR="#000000">Neither user can call out. There are no errors in the log that I can</FONT><BR>
<FONT COLOR="#000000">see.</FONT><BR>
<BR>
<FONT COLOR="#000000">What did I miss?</FONT><BR>
<BR>
<FONT COLOR="#000000">Thanks</FONT><BR>
<FONT COLOR="#000000">-- </FONT><BR>
<FONT COLOR="#000000">Bob Crandell</FONT><BR>
<FONT COLOR="#000000">Assured Computing, Inc.</FONT><BR>
<FONT COLOR="#000000">http://www.assuredcomp.com/</FONT><BR>
<FONT COLOR="#000000">541-868-0331</FONT><BR>
<FONT COLOR="#000000">ComputerBase</FONT><BR>
<FONT COLOR="#000000">http://www.computerbaseusa.com/</FONT><BR>
<FONT COLOR="#000000">541-349-0404</FONT><BR>
<BR>
<FONT COLOR="#000000">_______________________________________________</FONT><BR>
<FONT COLOR="#000000">--Bandwidth and Colocation Provided by http://www.api-digital.com--</FONT><BR>
<BR>
<FONT COLOR="#000000">asterisk-gui mailing list</FONT><BR>
<FONT COLOR="#000000">To UNSUBSCRIBE or update options visit:</FONT><BR>
<FONT COLOR="#000000"> http://lists.digium.com/mailman/listinfo/asterisk-gui</FONT><BR>
<BR>
</BLOCKQUOTE>
</BODY>
</HTML>