Hi Steve,<div><br></div><div>I am generating outbound calls to the PSTN network connected to my FreeSWITCH box via PRI line. Below is a piece of debug log for the same. There is some more but I am pasting the error lines out of that:</div>
<div><br></div><div><div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:3/9891605603</div><div>ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:3/9891605603 [ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5]</div>
<div>ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:3/9891605603) State Change CS_NEW -> CS_INIT</div><div>ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:3/9891605603 [BREAK]</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab444a42-9dbb-11e2-b0bc-6fb3d8274ed5 to channel 1:3</div><div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c3][1:3] Changed state from DOWN to DIALING</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables</div><div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[95]</div><div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1]</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9467797613]</div><div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en]</div><div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true]</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true]</div><div>2013-04-05 12:10:13.234533 [INFO] ftmod_zt.c:671 Setting echo cancel to 64 taps for 1:4</div><div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20ms</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/1:4/9467797613</div><div>ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [NOTICE] switch_channel.c:976 New Channel FreeTDM/1:4/9467797613 [ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5]</div>
<div>ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1251 (FreeTDM/1:4/9467797613) State Change CS_NEW -> CS_INIT</div><div>ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 2013-04-05 12:10:13.234533 [DEBUG] switch_core_session.c:1310 Send signal FreeTDM/1:4/9467797613 [BREAK]</div>
<div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1270 Attached session ab461d0e-9dbb-11e2-b0c0-6fb3d8274ed5 to channel 1:4</div><div>2013-04-05 12:10:13.234533 [DEBUG] mod_freetdm.c:1758 [s1c4][1:4] Changed state from DOWN to DIALING</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[96]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1]</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8802865008]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true]</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true]</div><div>ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW]</div>
<div>ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY</div><div>ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY</div>
<div>ab47f0c0-9dbb-11e2-b0c4-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep</div><div>2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION]</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[97]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[8826454579]</div>
<div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true]</div>
<div>ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW]</div><div>ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY</div>
<div>ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY</div><div>ab49c22e-9dbb-11e2-b0c8-6fb3d8274ed5 2013-04-05 12:10:13.254533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep</div>
<div>2013-04-05 12:10:13.254533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION]</div><div>2013-04-05 12:10:13.254533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]</div>
<div>2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables</div><div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [voiceMessageID]=[98]</div><div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [respreqd]=[1]</div>
<div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [mobnum]=[9868599566]</div><div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [lang]=[en]</div><div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true]</div>
<div>2013-04-05 12:10:13.274533 [DEBUG] switch_event.c:1608 Parsing variable [continue_on_fail]=[true]</div><div>ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [NOTICE] mod_freetdm.c:1766 Close Channel N/A [CS_NEW]</div>
<div>ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:568 () Running State Change CS_DESTROY</div><div>ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY</div>
<div>ab4b9752-9dbb-11e2-b0cc-6fb3d8274ed5 2013-04-05 12:10:13.274533 [DEBUG] switch_core_state_machine.c:578 (N/A) State DESTROY going to sleep</div><div>2013-04-05 12:10:13.274533 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [NORMAL_CIRCUIT_CONGESTION]</div>
<div>2013-04-05 12:10:13.274533 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]</div><div>2013-04-05 12:10:13.274533 [DEBUG] ftdm_state.c:541 [s1c1][1:1] Executing state processor for DIALING</div>
<div>2013-04-05 12:10:13.274533 [DEBUG] ftmod_libpri.c:935 -- 1:1 STATE [DIALING]</div><div>2013-04-05 12:10:13.274533 [DEBUG] mod_freetdm.c:2657 got clear channel sig [DIALING]</div><br><div class="gmail_quote">On Fri, Apr 5, 2013 at 12:33 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">NORMAL_CIRCUIT_CONGESTION is an error that can be returned by the SIP server you're calling out through. Show us the debug log and enable sip trace (sofia global siptrace on).<div>
<br></div><div>Almost certainly that error is coming from your SIP provider and you'll need to take it up with them to see why they're returning that.</div>
<div><br></div><div>Check for the SIP code in the final response they return (eg 503 Service Unavailable) and the Reason header in that that message (which is where you may see NORMAL_CIRCUIT_CONGESTION).</div>
<div><br></div><div>-Steve</div><div><div class="h5"><div><br><div class="gmail_extra"><br><br><div class="gmail_quote">On 5 April 2013 07:24, Ashish gautam <span dir="ltr"><<a href="mailto:ashish@nms.co.in" target="_blank">ashish@nms.co.in</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Hi Steven,<div><br></div><div>Thanks for responding.</div>
<div><br></div><div>I am checking the console at debug level and getting [NORMAL_CIRCUIT_CONGESTION] error. This must not happen as the number of calls I am generating is only 4. What is happening is, the ES generates calls to all the numbers except the last one (it always fails for the last number and shows NORMAL_CIRCUIT_CONGESTION). If the problem would be in originate dialstring, calls to other numbers would also be unsuccessful but they are working fine.</div>
<div><br></div><div>Thanks.</div><div>-Ashish </div><div><br><div class="gmail_quote">On Fri, Apr 5, 2013 at 12:17 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Send FreeSWITCH-users mailing list submissions to<br>
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<br>Today's Topics:<br>
<br>
1. Re: One of my extensions Caller ID says Anonymous! (Vik Killa)<br>
2. Re: freeswitch say application currency in multipal language<br>
(Michael Collins)<br>
3. Re: One of my extensions Caller ID says Anonymous! (Avi Marcus)<br>
4. Re: Call not originated through event socket (Steven Ayre)<br>
<br><br>---------- Forwarded message ----------<br>From: Vik Killa <<a href="mailto:vipkilla@gmail.com" target="_blank">vipkilla@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 4 Apr 2013 14:13:26 -0400<br>Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous!<br><div dir="ltr"><div>Please let us know if and how you get it resolved. I'll do the same if I can figure it out. Thanks.</div>
<div><br></div><snip></div>
<br><br>---------- Forwarded message ----------<br>From: Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 4 Apr 2013 11:14:52 -0700<br>Subject: Re: [Freeswitch-users] freeswitch say application currency in multipal language<br>I don't believe that there is currently a way to do this easily right now. We just spoke about languages on yesterday's conference call and this is a prime example of the kinds of things that we will need to overcome. <br>
<br>Additionally I don't believe that I have any currencies other than dollar.wav and dollars.wav for the English sounds. I'll be glad to get them ordered. Could the community at large send me some ideas for units of currency? Here are a few ideas:<br>
<br>euro, euros<br>franc, francs<br>Canadian, Australian, US dollar/dollars<br>pound, pounds<br><br>Send me some more ideas and I will get them added to the to-be-recorded list.<br><br>-MC<br><br><div class="gmail_quote">
On Thu, Apr 4, 2013 at 1:52 AM, bhavik patel <span dir="ltr"><<a href="mailto:bhavikpatel14388@gmail.com" target="_blank">bhavikpatel14388@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir="ltr">Hi all,
<br>I use free switch and i want to play sounds file like if user has
credit in USD then doller.wav file play and EUR then another file will
be play.
<br><br>Currently it play doller.wav by default in
/usr/local/freeswitch/sounds/currency/en/doller.wav but i want to play
EUR so how can this possible.
<br><br>Is that any easy way to do this thing in multi language currency play in say application.
<br><br>i use this syntax in my free-switch dial plan
<br>$dialstring = "<action application=\"say\" data=\"en CURRENCY PRONOUNCED $credit_balance\"/>";
<br><br>Thanks In advance...<span><font color="#888888"><br clear="all"><br>-- <br>Thanks,<br>Bhavik Patel<br><br>
</font></span></div><div>
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<br></div></blockquote></div><div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
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<br><br></div>---------- Forwarded message ----------<br>From: Avi Marcus <<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 4 Apr 2013 21:16:04 +0300<br>Subject: Re: [Freeswitch-users] One of my extensions Caller ID says Anonymous!<br><div dir="ltr">Indeed, <span style="font-family:arial,sans-serif;font-size:12.571428298950195px">outbound_caller_id_name is just a variable.</span><div>
<font face="arial, sans-serif"><span style="font-size:12.571428298950195px"><br>
</span></font><div><span style="font-family:arial,sans-serif;font-size:12.571428298950195px">To use it you have to tell the dialplan to set </span><span style="font-family:arial,sans-serif;font-size:12.571428298950195px">effective_caller_id_name=${</span><span style="font-family:arial,sans-serif;font-size:12.571428298950195px">outbound_caller_id_name}.</span></div>
<div><span style="font-family:arial,sans-serif;font-size:12.571428298950195px">effective_caller_id_name IS a variable that actually gets used.</span><span style="font-family:arial,sans-serif;font-size:12.571428298950195px"><br>
</span></div><div class="gmail_extra">(The default dialplan does this.)</div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif;font-size:small">-Avi Marcus</span></div>
</div><div><span style="font-family:Verdana,Arial,Helvetica,sans-serif;font-size:small">BestFone</span></div>
<br><br><div class="gmail_quote">On Thu, Apr 4, 2013 at 8:26 PM, Steven Schoch <span dir="ltr"><<a href="mailto:schoch+freeswitch.org@xwin32.com" target="_blank">schoch+freeswitch.org@xwin32.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">In my directory, I set the variable "effective_caller_id_name" instead of "internal_caller_id_name". (I also set the "outbound_caller_id_name" variable just like your example, although that never seems to get sent to the PSTN.)<span><font color="#888888"><span><font color="#888888"><div>
<br></div><div>-- </div><div>Steve</div></font></span></font></span></div><div>
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<br></div></blockquote></div><br></div></div></div><div><div>
<br><br>---------- Forwarded message ----------<br>From: Steven Ayre <<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 4 Apr 2013 19:46:22 +0100<br>Subject: Re: [Freeswitch-users] Call not originated through event socket<br><div dir="ltr">+1<div><br></div><div>'/log 9' and watch the debug output. It'll be very clear about what's happening. Most likely there's something wrong about your usage of originate.</div>
<div>
<br></div><div>-Steve</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 4 April 2013 19:08, Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">You could also just hop on fs_cli and watch the output while you make a test call with the event socket. Most likely you'll see what's going on.<br>
<br>-MC<br><br><div class="gmail_quote"><div><div>On Thu, Apr 4, 2013 at 10:39 AM, Hector Geraldino <span dir="ltr"><<a href="mailto:Hector.Geraldino@ipsoft.com" target="_blank">Hector.Geraldino@ipsoft.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div>
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<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">+OK doesn’t mean the call was generated, it just means that FreeSWITCH received (and processed) the command.<u></u><u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"><u></u> <u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">You should probably be better served by listening to the events for each one of these calls. There are a few handful events that can tell you if the channel
was created, the call was originated, if it was answered and when it was dropped with the drop reason.<u></u><u></u></span></p>
<p><span style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"><u></u> <u></u></span></p>
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<p><b><span style="font-size:10pt;font-family:Tahoma,sans-serif">From:</span></b><span style="font-size:10pt;font-family:Tahoma,sans-serif"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>On Behalf Of </b>Ashish gautam<br>
<b>Sent:</b> Thursday, April 04, 2013 5:26 AM<br>
<b>To:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Subject:</b> [Freeswitch-users] Call not originated through event socket<u></u><u></u></span></p>
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<p><u></u> <u></u></p>
<p>Hi,<u></u><u></u></p>
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<p><u></u> <u></u></p>
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<p>I am generating calls through event socket using originate action to multiple numbers to PSTN network. ES is returning '+OK' for the call not generated. Still the call is not generated.<u></u><u></u></p>
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<p><u></u> <u></u></p>
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<p>Any help is appreciated.<u></u><u></u></p>
</div>
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<p><u></u> <u></u></p>
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<p>Thanks.<u></u><u></u></p>
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<p><u></u> <u></u></p>
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<p>-Ashish<u></u><u></u></p>
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</div>
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