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Hi,<BR>
<BR>
I have since reinstalled this a few times trying to understand relationships, permissions and such like. Now I'm ready to dial out again. This time there is an error: Call failed: 603 Declined<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 = 13857)<BR>
Verbosity is at least 6<BR>
<BR>
-- Executing [96899159@DLPN_DialPlan1:1] Macro("SIP/6002-08227968", "trunkdial-failover-0.3|Zap/g1/5551212||trunk_1|trunk_1") in new stack<BR>
-- Executing [s@macro-trunkdial-failover-0.3:1] Set("SIP/6002-08227968", "CALLERID(num)=") in new stack<BR>
-- Executing [s@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6002-08227968", "0?1-dial|1") in new stack<BR>
-- Executing [s@macro-trunkdial-failover-0.3:3] Set("SIP/6002-08227968", "CALLERID(all)=") in new stack<BR>
-- Executing [s@macro-trunkdial-failover-0.3:4] Goto("SIP/6002-08227968", "1-dial|1") in new stack<BR>
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)<BR>
-- Executing [1-dial@macro-trunkdial-failover-0.3:1] Dial("SIP/6002-08227968", "Zap/g1/5551212") in new stack<BR>
[Feb 27 11:58:07] WARNING[29008]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)<BR>
== Everyone is busy/congested at this time (1:0/0/1)<BR>
-- Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf("SIP/6002-08227968", "0 > 0 ?1-CHANUNAVAIL|1:1-out|1") in new stack<BR>
-- Goto (macro-trunkdial-failover-0.3,1-out,1)<BR>
-- Executing [1-out@macro-trunkdial-failover-0.3:1] Hangup("SIP/6002-08227968", "") in new stack<BR>
== Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6002-08227968' in macro 'trunkdial-failover-0.3'<BR>
== Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited non-zero on 'SIP/6002-08227968'<BR>
<BR>
This is a test box. I'm the only one on it.<BR>
One Digium TDM400P with one FXO and one FXS daughter card.<BR>
<BR>
Thanks<BR>
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Bob Crandell
Assured Computing, Inc.
<A HREF="http://www.assuredcomp.com/">http://www.assuredcomp.com/</A>
541-868-0331
ComputerBase
<A HREF="http://www.computerbaseusa.com/">http://www.computerbaseusa.com/</A>
541-349-0404
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<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>
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<FONT COLOR="#000000">To UNSUBSCRIBE or update options visit:</FONT><BR>
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