The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI. there is some misconfiguration in FreePBX and your dialled number is not hitting any dial-able rule. See your FreePBX guide. <div><br><br><div class="gmail_quote">
On Thu, Sep 29, 2011 at 11:01 AM, michael k <span dir="ltr"><<a href="mailto:michael@inapp.com">michael@inapp.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hi,<br><br> Please see the sample. <br><br><h2>A ) Analog Hardware</h2><table border="0" cellpadding="5" cellspacing="1"><thead><tr><th>Type</th>
<th>Ports</th>
<th>Action</th>
</tr>
</thead>
<tbody>
<tr>
<td>FXO Ports</td>
<td>1</td>
<td><a href="http://192.168.1.134/admin/config.php?type=setup&display=dahdi&dahdi_form=analog_signalling&ports=fxo" target="_blank">Edit</a></td>
</tr>
<tr>
<td>FXS Ports</td>
<td>--</td>
<td><br></td></tr></tbody></table><br><h2>B) Analog FXO Ports</h2>Port 1: <b>Loop Start</b>
Group: <b>0</b>
Context:
<b> from-analog</b><br><br><font size="4"><b><br>C ) ZAP Trunk (DAHDI compatibility Mode)</b></font><br><br><br><a>Trunk Description</a>:
<input size="30" name="trunk_name" value="ZAP Channel g0" type="text"><br><table><tbody><tr><td><a>Outbound Caller ID</a>:
</td><td>
<input size="30" name="outcid" value="04712527271" type="text">
</td>
</tr>
<tr>
</tr><tr>
<td>
<a>CID Options</a>:
</td><td>
<br></td>
</tr>
<tr>
<td>
<a>Maximum Channels</a>:
</td><td>
<input size="3" name="maxchans" value="1" type="text">
</td>
</tr>
<tr>
<td><a>Disable Trunk</a>:
</td>
<td>
<input name="disabletrunk" type="checkbox"><small>Disable</small>
</td>
</tr>
<tr>
<td><a>Monitor Trunk Failures</a>:
</td>
<td>
<input disabled style="background:#DDD" size="20" name="failtrunk" value="" type="text">
<input name="failtrunk_enable" value="1" type="checkbox"><small>Enable</small>
</td>
</tr>
<tr>
<td colspan="2">
<h4>Outgoing Dial Rules</h4>
</td>
</tr>
<tr>
<td valign="top">
<a>Dial Rules</a>:
</td><td valign="top">
<textarea cols="20" rows="5" name="dialrules">0471+NXXXXXX</textarea><br>
</td>
</tr>
<tr>
<td>
<a>Dial Rules Wizards</a>:
</td><td valign="top">
<br></td>
</tr>
<tr>
<td>
<a>Outbound Dial Prefix</a>:
</td><td>
<input size="8" name="dialoutprefix" value="" type="text">
</td>
</tr>
<tr>
<td colspan="2">
<h4>Outgoing Settings</h4>
</td>
</tr>
<tr>
<td>
<a>Zap Identifier (trunk name)</a>:
</td><td>
<input size="8" name="channelid" value="g0" type="text">
</td>
</tr>
<tr>
<td colspan="2">
<br></td></tr></tbody></table><br><br><font size="4"><b>D ) INBOUND route </b></font><br><br> <a>Description</a>:
<input name="description" value="catch all" type="text"><br><input name="goto0" value="Extensions" checked type="radio"> <label>Extensions:</label> 199<br>
<font size="4"><b><br><br>E ) </b><b>OUTBOUND Route</b></font><br><br><table><tbody><tr><td><a>Route Name</a>:
</td>
<td>
9_outside
</td>
</tr>
<tr>
<td><a>Route CID:</a></td>
<td>
<input size="20" name="routecid" value="" type="text">
<input name="routecid_mode" value="override_extension" type="checkbox"><small>Override Extension CID</small>
</td>
</tr>
<tr>
<td><a>Route Password:</a></td>
<td><input size="20" name="routepass" value="" type="text"></td>
</tr>
<tr>
<td><a>PIN Set</a>:</td>
<td>
<br></td>
</tr>
<tr>
<td><a>Emergency Dialing</a>:</td>
<td><input name="emergency" value="yes" type="checkbox"></td>
</tr>
<tr>
<td><a>Intra Company Route</a>:</td>
<td><input name="intracompany" value="yes" type="checkbox"></td>
</tr>
<tr>
<td><a>Music On Hold?</a></td>
<td>
<br></td>
</tr>
<tr>
<td colspan="2">
<a>Dial Patterns</a>
</td>
</tr>
<tr>
<td>
<br></td><td>
<textarea cols="20" rows="5" name="dialpattern">8|NXXNXXXXXX
8|NXXXXXX</textarea><br>
</td>
</tr>
<tr>
<td>
<a>Dial patterns wizards</a><strong>:
</strong></td><td>
<br></td>
</tr>
<tr>
<td colspan="2">
<a>Trunk Sequence ZAP/g0</a>
</td>
</tr>
<tr>
<td align="right">0
</td>
<td>
<img src="http://192.168.1.134/admin/images/trash.png" style="float:none;margin-left:0px;margin-bottom:0px" title="Click here to remove this trunk">
<img src="http://192.168.1.134/admin/images/blank.gif" style="float:none;margin-left:0px;margin-bottom:0px" height="11" width="9"></td></tr></tbody></table><br><font size="4"><b><br>F ) In command Line I can see the following things </b></font><br>
<br><br>[root@astrisks ~]# <font size="4"><b>dahdi_cfg -vv</b></font><br><br><br>DAHDI Tools Version - 2.3.0<br><br>DAHDI Version: 2.3.0.1<br>Echo Canceller(s): <br>Configuration<br>======================<br><br><br>Channel map:<br>
<br>Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)<br><br>1 channels to configure.<br><br>Setting echocan for channel 1 to none<br><br><br>[root@astrisks ~]# <font size="4"><b>dahdi_scan</b></font><br>
<br>[1]<br>active=yes<br>alarms=OK<br>description=Wildcard X100P Board 1<br>name=WCFXO/0<br>manufacturer=Digium<br>devicetype=Wildcard X100P<br>location=PCI Bus 02 Slot 02<br>basechan=1<br>totchans=1<br>irq=193<br>type=analog<br>
port=1,FXO<br><br><br><br><font size="4"><u><b>Asterisk CLI</b></u></font> <br><br><br><b>astrisks*CLI> dahdi show status</b><br>
<br>
Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO<br>
Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)<br><br><font size="4"><b><br>output when i dialing to a local number</b></font><br><br>Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)<br>
Verbosity is at least 3<br> == Using SIP RTP TOS bits 184<br> == Using SIP RTP CoS mark 5<br> -- Executing [s@from-internal:1] Macro("SIP/199-0000003a", "hangupcall") in new stack<br> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-0000003a", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-0000003a", "1?skipblkvm") in new stack<br> -- Goto (macro-hangupcall,s,7)<br> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-0000003a", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br> -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-0000003a", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-0000003a' in macro 'hangupcall'<br>
== Spawn extension (from-internal, s, 1) exited non-zero on 'SIP/199-0000003a'<br> -- Executing [h@from-internal:1] Macro("SIP/199-0000003a", "hangupcall") in new stack<br> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-0000003a", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-0000003a", "1?skipblkvm") in new stack<br> -- Goto (macro-hangupcall,s,7)<br> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-0000003a", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br> -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-0000003a", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/199-0000003a' in macro 'hangupcall'<br>
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/199-0000003a'<div><div></div><div class="h5"><br><br><br><table border="0" cellpadding="5" cellspacing="1"><thead><tr><th><br></th><th><br>
</th><th><br></th></tr></thead><tbody><tr><td><br></td><td><br></td><td><br></td></tr><tr><td><br></td><td><br></td>
</tr></tbody></table><br><br><br><br><br><br><div class="gmail_quote">On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Some CLI logs will get you better help on the issue ! also paste the FXO configurations and how you configured it !<br><br><div class="gmail_quote"><div><div></div><div>On Wed, Sep 28, 2011 at 2:11 PM, michael k <span dir="ltr"><<a href="mailto:michael@inapp.com" target="_blank">michael@inapp.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div>Hi All,<br> <br> I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. <br>
<br>1. OUTBOUND caller id and Dialing rules in Freepbx. <br><br>2. INBOUND route <br><br>When i call to the PSTN number before connecting to the FXO card, i am getting a ringing. But i get a message like the "number is out of order" when i just connect the line to FXO card. <br>
<br>Please some one help me to resolve his issue <br>
<br></div></div>--<br>
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