<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I noticed tonegroup=es. What country are you in and do you know what method they use to do dtmf. Most likely we need a small tweak to set the dtmf method for your country.<div><br></div><div>Mike</div><div><br><div><div>On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><br>number => 1<br><br>This value should be set to the DID of the FXO line.<br>That way when a call hits FS it will go to that extension in the dialplan.<br>This is unrelated to callerid, it's the destination not the source.<br> <br>If the line has caller-id it will also be available when it's collected after the 2nd ring.<br><br><br><br><div class="gmail_quote">On Wed, Jan 14, 2009 at 5:54 AM, Tomás <span dir="ltr"><<a href="mailto:tomasborrella@gmail.com">tomasborrella@gmail.com</a>></span> wrote:<br> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi all,<br><br>I'm a new FreeSwitch user and this is my first email to the list.<br><br> I'm trying to configure my Home PBX with a Wildcard X101P (configured as FXO) and I have a problem receiving the caller/called ID from PSTN.<br> <br>This is the content of file "openzap.conf":<br><br>[span zt]<br>name => OpenZAP<br>number => 1<br>fxo-channel => 1<br><br>And this is the content of file "openzap.conf.xml":<br><br><configuration name="openzap.conf" description="OpenZAP Configuration"><br> <settings><br> <param name="debug" value="0"/><br> </settings><br> <analog_spans><br> <span id="1"><br> <param name="tonegroup" value="es"/><br> <param name="digit-timeout" value="2000"/><br> <param name="max-digits" value="11"/><br> <param name="dialplan" value="XML"/><br> <param name="context" value="default"/><br> <param name="enable-callerid" value="true"/><br> </span><br> </analog_spans><br></configuration><br> <br>As you can see the param "enable-callerid" is set to "true", but when I received and incoming call, FreeSwitch doesn't get neither the caller number nor the called number (instead of my home number, I receive a number 1, as can be seen on the following log of an incoming call):<br> <br>2009-01-14 20:43:12 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [941d6234-e273-11dd-bcdf-89190a30fe61]<br>2009-01-14 20:43:12 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing OpenZAP->1 in context default<br> 2009-01-14 20:43:12 [NOTICE] switch_ivr.c:1255 switch_ivr_session_transfer() Transfer OpenZAP/1:1/1 to enum[1@default]<br>2009-01-14 20:43:16 [INFO] switch_core_state_machine.c:122 switch_core_standard_on_routing() No Route, Aborting<br> 2009-01-14 20:43:16 [NOTICE] switch_core_state_machine.c:123 switch_core_standard_on_routing() Hangup OpenZAP/1:1/1 [CS_ROUTING] [NO_ROUTE_DESTINATION]<br>2009-01-14 20:43:16 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 (OpenZAP/1:1/1 ) Ended<br> 2009-01-14 20:43:16 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel OpenZAP/1:1/1 [CS_HANGUP]<br>2009-01-14 20:43:23 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel OpenZAP/1:1/1 [9af53280-e273-11dd-bcdf-89190a30fe61]<br> <br></blockquote></div></blockquote></div></div></body></html>