<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">What's the config for the 2951 look like with respect to that FXO port? The FXS is just going to signal the FXO to ring, it isn't capable of sending digits across (except caller-id if enabled). If there is an inbound dial-peer on the 2951 I'd expect you to get secondary dialtone as mentioned, if there's a connection-plar configured then an outbound call will be initiated to that destination and used to connect to the incoming call on the FXO port.<div><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">-Ryan</span>
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<br><div><div>On Feb 27, 2013, at 12:25 PM, Jeffrey Girard <<a href="mailto:jeffrey.girard@girardinc.com">jeffrey.girard@girardinc.com</a>> wrote:</div><br class="Apple-interchange-newline"><div fpstyle="1" ocsi="0" style="font-family: 'Lucida Grande'; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div style="direction: ltr; font-family: Tahoma; font-size: 10pt; "><div style="margin-top: 0px; margin-bottom: 0px; ">Thanks for the reply, but unfortunately, I dont think that is correct.</div><p style="margin-top: 0px; margin-bottom: 0px; "> </p><div style="margin-top: 0px; margin-bottom: 0px; "><a href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetoll.html">http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetoll.html</a></div><p style="margin-top: 0px; margin-bottom: 0px; "> </p><div style="margin-top: 0px; margin-bottom: 0px; ">On page 497, under the heading of Blocking Two-stage Dialing Service on Analog and Digital FXO Ports</div><p style="margin-top: 0px; margin-bottom: 0px; "> </p><h3 class="p_H_Head3">Blocking Two-stage Dialing Service on Analog and Digital FXO Ports</h3><a name="wp1068968"></a><div style="margin-top: 0px; margin-bottom: 0px; ">Cisco Unified CME 8.1 and later versions block the two-stage dialing service which is initiated when an Analog or Digital FXO port goes offhook and the private line automatic ringdown (PLAR) connection is not setup from the voice-port. As a result, no outbound dial-peer is selected for an incoming analog or digital FXO call and no dialed digits are collected from an FXO call. Cisco Unified CME and voice gateways disconnect the FXO call with cause-code "unassigned-number (1)". Cisco Unified CME uses the<span style="font-style: normal; font-weight: bold; "><span class="Apple-converted-space"> </span>no secondary dialtone</span><span class="Apple-converted-space"> </span>command by default from FXO voice-port to block the two-stage dialing service on Analog or digital FXO ports.</div><div style="font-family: 'Times New Roman'; font-size: 16px; "> </div><div style="font-family: 'Times New Roman'; font-size: 16px; ">This implies that if I do not configure connection plar on the FXO port, I should get secondary dialtone. This is what I always thought to be true.</div><div style="font-family: 'Times New Roman'; font-size: 16px; "> </div><div style="font-family: 'Times New Roman'; font-size: 16px; ">This document did, however, lead me to discover that the "no secondary dialtone" command was on by default on the FXO port. I reversed the command, but still no secondary dial-tone.</div><div style="font-family: 'Times New Roman'; font-size: 16px; "> </div><div style="font-family: 'Times New Roman'; font-size: 16px; ">I am going to do some more testing, which will include replacing the FXO and FXS cards</div><div style="font-family: 'Times New Roman'; font-size: 16px; "> </div><div style="font-family: 'Times New Roman'; font-size: 16px; ">Anyone else with suggestions?</div><div style="font-family: 'Times New Roman'; font-size: 16px; "> </div><div style="font-family: 'Times New Roman'; font-size: 16px; ">Jeff</div><div style="font-family: 'Times New Roman'; font-size: 16px; "><hr tabindex="-1"><div id="divRpF751539" style="direction: ltr; "><font size="2" face="Tahoma"><b>From:</b><span class="Apple-converted-space"> </span>ccieid1ot [<a href="mailto:ccieid1ot@gmail.com">ccieid1ot@gmail.com</a>]<br><b>Sent:</b><span class="Apple-converted-space"> </span>Tuesday, February 26, 2013 1:49 PM<br><b>To:</b><span class="Apple-converted-space"> </span>Jeffrey Girard<br><b>Cc:</b><span class="Apple-converted-space"> </span><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br><b>Subject:</b><span class="Apple-converted-space"> </span>Re: [cisco-voip] 2 stage inbound to FXO port issues<br></font><br></div><div></div><div>For calls inbound to FXO ports, you would need to configure connection plar.<br><br><div class="gmail_quote">On Mon, Feb 25, 2013 at 10:38 AM, Jeffrey Girard<span class="Apple-converted-space"> </span><span dir="ltr"><<a href="mailto:jeffrey.girard@girardinc.com" target="_blank">jeffrey.girard@girardinc.com</a>></span><span class="Apple-converted-space"> </span>wrote:<br><blockquote class="gmail_quote" style="border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid; margin: 0px 0px 0px 0.8ex; padding-left: 1ex; "><div><div style="font-family: Tahoma; direction: ltr; font-size: 10pt; "><div style="margin-top: 0px; margin-bottom: 0px; ">Simplified diagram is attached. Lab network.</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">Scenario: Using a 2811 router as a PSTN emulator. Has a 2 port FXS card installed. One port has an analog phone attached, other port is connected to a 4 port FXO card installed into a 2951 router. 2951 is H323 with CUCM 8.6. Simple/test dial peers configured. Snippets below:</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">PSTN Emulator:</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/0/0</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/0/1</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">dial-peer voice 1 pots</div><div style="margin-top: 0px; margin-bottom: 0px; ">description Emulating legacy PATCOM number</div><div style="margin-top: 0px; margin-bottom: 0px; ">destination-pattern 0999</div><div style="margin-top: 0px; margin-bottom: 0px; ">port 0/0/1</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">dial-peer voice 2 pots</div><div style="margin-top: 0px; margin-bottom: 0px; ">description Emulating tie line to Access Terminal</div><div style="margin-top: 0px; margin-bottom: 0px; ">destination-pattern 9999</div><div style="margin-top: 0px; margin-bottom: 0px; ">port 0/0/0</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">H323 Gateway Router</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/1/0</div><div style="margin-top: 0px; margin-bottom: 0px; ">trunk-group FXO</div><div style="margin-top: 0px; margin-bottom: 0px; ">cptone PK</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/1/1</div><div style="margin-top: 0px; margin-bottom: 0px; ">trunk-group FXO</div><div style="margin-top: 0px; margin-bottom: 0px; ">cptone PK</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/1/2</div><div style="margin-top: 0px; margin-bottom: 0px; ">trunk-group FXO</div><div style="margin-top: 0px; margin-bottom: 0px; ">cptone PK</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-port 0/1/3</div><div style="margin-top: 0px; margin-bottom: 0px; ">trunk-group FXO</div><div style="margin-top: 0px; margin-bottom: 0px; ">cptone PK</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">dial-peer voice 1 pots</div><div style="margin-top: 0px; margin-bottom: 0px; ">description Incoming Call Routing</div><div style="margin-top: 0px; margin-bottom: 0px; ">incoming called-number .</div><div style="margin-top: 0px; margin-bottom: 0px; ">direct-inward-dial</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">dial-peer voice 2 voip</div><div style="margin-top: 0px; margin-bottom: 0px; ">description Incoming Call Routing</div><div style="margin-top: 0px; margin-bottom: 0px; ">incoming called-number .</div><div style="margin-top: 0px; margin-bottom: 0px; ">voice-class codec 1 </div><div style="margin-top: 0px; margin-bottom: 0px; ">dtmf-relay h245-alphanumeric</div><div style="margin-top: 0px; margin-bottom: 0px; ">no vad</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!<br>voice-port 0/1/0<br>trunk-group FXO<br>cptone PK<br>!<br>voice-port 0/1/1<br>trunk-group FXO<br>cptone PK<br>!<br>voice-port 0/1/2<br>trunk-group FXO<br>cptone PK<br>!<br>voice-port 0/1/3<br>trunk-group FXO<br>cptone PK<br>!</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><div style="margin-top: 0px; margin-bottom: 0px; ">dial-peer voice 999 pots</div><div style="margin-top: 0px; margin-bottom: 0px; ">description Dummy dial peer to test BYOPBX</div><div style="margin-top: 0px; margin-bottom: 0px; ">destination-pattern 0999</div><div style="margin-top: 0px; margin-bottom: 0px; ">port 0/1/0</div><div style="margin-top: 0px; margin-bottom: 0px; ">forward-digits all</div><div style="margin-top: 0px; margin-bottom: 0px; ">!</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">I have other dial-peers on this router (not shown) that will take the inbound digits. However, Im never able to pass the digits.</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">Calls from the VoIP network to the simulated PSTN work fine. Calls from the PSTN to the VoIP network fail. </div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">When I pick up the analog phone, I get the expected dial tone. I enter "9999" and the call rings to the FXS port and I hear the FXO port answer. I get the expected secondary (2 stage) dial tone, but then before I can press any digits, the FXo port hangs up and I am left with the PSTN router dial tone.</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">debug voip dialpeer on the H323 router shows no results at all when I try to place the call.</div><div style="margin-top: 0px; margin-bottom: 0px; ">debug voip dialpeer on the PSTN router shows the expected dial peer matching and the expected result. File is attached.</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">Additionally, when I observe the output of show voice port summary while I place the call, I see the FXS port (the one that is tied to the FXO port) go off hook and then go back on hook (file also attached)</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">So, the FXO port appears to be looking for something from the FXS when it answers the call, doesnt get it (or doesnt like what it gets) so it hangs up.</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">Looking for help...</div><p style="margin-top: 0px; margin-bottom: 0px; padding: 0px; min-height: 8pt; "><br class="webkit-block-placeholder"></p><div style="margin-top: 0px; margin-bottom: 0px; ">Jeff</div></div></div><br>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br><br></blockquote></div><br><br clear="all"><br>--<span class="Apple-converted-space"> </span><br>duy<br>CCIE #27737 Voice<br></div></div></div>_______________________________________________<br>cisco-voip mailing list<br><a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>https://puck.nether.net/mailman/listinfo/cisco-voip</div></div><br></div></body></html>