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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>You can get the uuid in fs_cli by using the show channels, or show calls commands. Also, the parameter for the eavesdrop app goes between parenthesis, so it’s &eavesdrop(uuid) <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Finally, try to do the originate without any additional arguments, just:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> originate originateString<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>and if it works then add the &eavesdrop(uuid). It will help you to figure out where the problem might be.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Liu, Xingkun<br><b>Sent:</b> Tuesday, February 28, 2012 6:14 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] How to bridge a call to an extensiondefined in dialplan<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p style='margin-bottom:12.0pt'><span style='font-size:10.0pt'>From the fs_cli command line, I don't know where to get the UUID for eavesdrop.<br>Anyway use "all" instead of a specific UUID for eavesdrop.<br><br>It looks like the same results as I issue the command from the ESL app.<br><br>From fs_cli command line, I issue:<br><br>originate {ignore_early_media=true}sofia/internal/1003%192.168.0.2 &eavesdrop all<br><br>I got same error: [ERR] mod_dptools.c:714 Usage: [all | <uuid>]<br><br>When I issue:<br><br>originate {ignore_early_media=true}sofia/internal/1003%192.168.0.2 &socket(127.0.0.1:9981 async full) &eavesdrop all<br><br>the softphone seems to work fine, I clicked Answer button, but still could not hear any thing from the speaker<br>when I make a call to my app.<br><br><br>By the way, which version of FS are you using, the latest Git version?<br><br> ( I am afraid I don't have time to try the latest FS version tonight, may try it later on)<br><br><br><br>-----Original Message-----<br>From: freeswitch-users-bounces@lists.freeswitch.org on behalf of Hector Geraldino<br>Sent: Tue 2/28/2012 22:30<br>To: FreeSWITCH Users Help<br>Subject: Re: [Freeswitch-users] How to bridge a call to an extensiondefined in dialplan<br><br>Sorry, it works for me.<br><br>Try to do it from the fs_cli command line and share the results.<br><br>From: freeswitch-users-bounces@lists.freeswitch.org [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Liu, Xingkun<br>Sent: Tuesday, February 28, 2012 5:11 PM<br>To: FreeSWITCH Users Help<br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br><br><br>Yeah, I tried this syntax before. I didn't work.<br>Now tried again, I got error as below and the softphone(1003) was terminated immediately:<br><br>2012-02-28 21:52:20.227899 [INFO] switch_channel.c:2695 sofia/internal/1003 Flipping CID from "" <0000000000> to "Outbound Call" <1003%192.168.0.2><br>2012-02-28 21:52:20.227899 [ERR] mod_dptools.c:714 Usage: [all | <uuid>]<br>2012-02-28 21:52:20.227899 [NOTICE] switch_core_state_machine.c:226 sofia/internal/1003 has executed the last dialplan instruction, hanging up.<br><br><br>And according to the wiki, the originate api needs a Call_from parameter:<br><br>Usage: originate <call_url> <exten><br> <call_url> URL you are calling.<br> <exten> Extension you are calling from<br><br>Or is this because I have a bit old version of FreeSwitch?<br> FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600)<br><br><br>Best,<br>Xing<br><br><br>-----Original Message-----<br>From: freeswitch-users-bounces@lists.freeswitch.org on behalf of Hector Geraldino<br>Sent: Tue 2/28/2012 20:33<br>To: FreeSWITCH Users Help<br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br>Why are you specifying the 5801 extension before the eavesdrop? You want to dial the Softphone extension: 1003, so all you need to do is to specifiy it on the originate string and pass the control to the eavesdrop app:<br><br>originate sofia/internal/1003@X.X.X.X<<a href="mailto:sofia/internal/1003@X.X.X.X">mailto:sofia/internal/1003@X.X.X.X</a>> &eavesdrop(uuid)<br><br>You don't need any more references to 5801, specially because you already have the UUID which is all you need to do the eavesdrop<br><br>From: freeswitch-users-bounces@lists.freeswitch.org [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of x.liu<br>Sent: Tuesday, February 28, 2012 2:27 PM<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br>Thanks again!<br><br>Yes, I used dvarne's Java ESL library for a while.<br><br>It works fine for the recording the a-leg, but the b-leg which is originated with eavesdrop to a Softphone does not work as expected.<br>There was an error in my originate usage:<br><br><br>originate sofia/internal/1003%x.x.x.x 5801&eavesdrop " + this.channelUniqueID);<br>I changed the extension 5801 to &socket(127.0.0.1:9981 async full) which is the action inside the extension 5801<br>then my softphone does not hang up any more, but I still can not hear the conversation of a-leg on the softphone.<br><br>There may be something wrong with my use of the combination of originate and eavesdrop.<br>I will look into it again.<br><br>Best,<br>Xing<br><br><br>On 02/28/2012 06:21 PM, Hector Geraldino wrote:<br><br>Ok, I can see you're using dvarne's java ESL library. Please download the source code from GIT and try to run the outbound client demo (located in src/test/java/SocketClientTest.java)<br><br><br><br>Assuming you're already familiar with the netty programming model, here's what I did: I changed the implementation of the SimpleHangupOutboundHandler class in the demo to do what you're asking: it will first answer the call, bridge it to an extension, record the call and make a 2nd call to another extension using the combination of originate + eavesdrop. Backup the original file and replace it with this new implementation. Also adjust the port number on the SocketClientTest class to match the one configured in your dialplan.<br><br><br><br>Hope this serve as an starting point...<br><br><br><br>-----Original Message-----<br><br>From: freeswitch-users-bounces@lists.freeswitch.org<<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of x.liu<br><br>Sent: Tuesday, February 28, 2012 12:40 PM<br><br>To: freeswitch-users@lists.freeswitch.org<<a href="mailto:freeswitch-users@lists.freeswitch.org">mailto:freeswitch-users@lists.freeswitch.org</a>><br><br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br><br><br>Thanks for the detailed explanations, Hector!<br><br><br><br>First, sorry for my further questions here:<br><br> In 2. you originate a 2nd call to an internal extension, what is this<br><br>internal extension doing?<br><br> In 5. from where you originate the 3rd call, your java app or<br><br>dialplan, or does it matter?<br><br> and to which extension?<br><br><br><br>I am asking those questions because I haven't got mine work and want to<br><br>fully understand how you did it.<br><br><br><br>I originate a 2rd call to my softphone from my java app. The softphone<br><br>was ringing and I clicked Answer button on the GUI<br><br>then it was immediately terminated.<br><br><br><br>I got some confusing outputs like<br><br> Hangup sofia/internal/1003 [CS_ROUTING] [NO_ROUTE_DESTINATION] and<br><br> Transfer sofia/internal/1003 to<br><br>&eavesdrop[1003@b38cb4c6-620e-11e1-8992-0b1a37fbec21]<br><br><br><br>I copy full outputs here:<br><br><br><br>2012-02-28 13:18:28.056043 [NOTICE] switch_channel.c:920 New Channel<br><br>sofia/internal/1003 [b3a3b61c-620e-11e1-8995-0b1a37fbec21]<br><br>2012-02-28 13:18:28.076030 [NOTICE] mod_dptools.c:1117 Channel<br><br>[sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>>] has been answered<br><br>2012-02-28 13:18:29.098072 [NOTICE] sofia.c:5460 Ring-Ready<br><br>sofia/internal/1003!<br><br>2012-02-28 13:18:31.839862 [NOTICE] sofia.c:6077 Channel<br><br>[sofia/internal/1003] has been answered<br><br>2012-02-28 13:18:31.839862 [NOTICE] switch_core_state_machine.c:166<br><br>Hangup sofia/internal/1003 [CS_ROUTING] [NO_ROUTE_DESTINATION]<br><br>2012-02-28 13:18:31.839862 [NOTICE] switch_ivr.c:1711 Transfer<br><br>sofia/internal/1003 to &eavesdrop[1003@b38cb4c6-620e-11e1-8992-0b1a37fbec21]<br><br>2012-02-28 13:18:31.839862 [NOTICE] switch_core_session.c:1395 Session<br><br>16 (sofia/internal/1003) Ended<br><br>2012-02-28 13:18:31.839862 [NOTICE] switch_core_session.c:1397 Close<br><br>Channel sofia/internal/1003 [CS_DESTROY]<br><br>2012-02-28 13:19:02.756130 [NOTICE] sofia.c:634 Hangup<br><br>sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>> [CS_EXECUTE] [NORMAL_CLEARING]<br><br><br><br><br><br>Any more thoughts?<br><br><br><br>Desperately want to sort this out for tomorrow morning's meeting!<br><br><br><br>Thanks!<br><br>Xing<br><br><br><br><br><br><br><br><br><br>On 02/28/2012 03:24 PM, Hector Geraldino wrote:<br><br>You can find the syntax for eavesdrop in the wiki. Here's how I have it working:<br><br><br><br> api originate sofia/internal/1234@x.x.x.x&eavesdrop<<a href="mailto:sofia/internal/1234@x.x.x.x&eavesdrop">mailto:sofia/internal/1234@x.x.x.x&eavesdrop</a>>(aleg_uuid)<br><br><br><br>You can try this command in the fs_cli command line (w/o the 'api' prefix). Eavesdrop will allow you to jump into a call and monitor the conversation. If the call has been bridged, you can hear both legs.<br><br><br><br>You can do the bridge however you want. The bridge operation is not related to the eavesdrop, so the way I have this working is:<br><br><br><br>1. I answer the call in the dialplan, and transfer the control to my java app (just as you're doing right now)<br><br>2. I originate a 2nd call to an internal extension.<br><br>3. When the 2nd call is answered, I bridge both calls using uuid_bridge uuid_lega uuid_legb (you can even do the bridge from the dialplan)<br><br>4. I start recording the call using 'uuid_record uuid_lega' (again, you can call the record application from the dialplan)<br><br>5. I can (if desired) monitor the call by originating a 3rd call to the manager extension + passing the uuid_lega to the eavesdrop app, as discussed before.<br><br><br><br>Hope this helps<br><br><br><br><br><br>-----Original Message-----<br><br>From: freeswitch-users-bounces@lists.freeswitch.org<<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of x.liu<br><br>Sent: Tuesday, February 28, 2012 9:08 AM<br><br>To: freeswitch-users@lists.freeswitch.org<<a href="mailto:freeswitch-users@lists.freeswitch.org">mailto:freeswitch-users@lists.freeswitch.org</a>><br><br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br><br><br>Hi Hector,<br><br><br><br>Many thanks for your advice!<br><br><br><br>I had a try but without success yet. What I am doing is<br><br><br><br>1. call to an extension: 5801, it will execute my ESL Java app. I call<br><br>it a-leg<br><br><extension name="myEslNew5801"><br><br><condition field="destination_number" expression="^5801$"><br><br><action application="limit" data="hash outgoing 5801 4" /><br><br><action application="socket" data="127.0.0.1:9981 async full"/><br><br></condition><br><br></extension><br><br><br><br>2. From my ESL Java app, I originate b-leg to my softphoe and eavesdrop<br><br>on extension 5801 by<br><br> response = sendSyncSingleLineCommand(ctx.getChannel(), "api<br><br>originate sofia/internal/1003%x.x.x.x 1003&eavesdrop " +<br><br>this.channelUniqueID);<br><br> the channelUniqueID is the UUID of my a-leg session.<br><br><br><br> So I hope my app do the recording and talking to the user and I<br><br>listen it from the softphone 1003 via eavesdrop.<br><br><br><br> At the moment I got message: Hangup on 1003 with [CS-ROUTING]<br><br>[NO_ROUTE_DESTINATION]<br><br><br><br> I am not quite sure if I understand correctly how the eavesdrop works:<br><br><br><br> Which UUID should I use in "&eavesdrop UUID", the existing one of<br><br>a-leg or I need to create a new one for b-leg?<br><br> Do I need to do the bridge, in extension 5801 or in my app?<br><br><br><br><br><br>Thanks!<br><br>Xing<br><br><br><br><br><br><br><br>On 02/27/2012 08:59 PM, Hector Geraldino wrote:<br><br>You can: bridge two endpoints (A -> B), record the conversation using uuid_record (or the record application), and have a 3rd extension hearing the conversation using the eavesdrop app. Here's how I've done this before:<br><br><br><br>Incoming call (A-Party)<br><br>Bridge call to B-Party (bridge 1xxx@example.domain<<a href="mailto:1xxx@example.domain">mailto:1xxx@example.domain</a>>)<br><br>Record the conversation (uuid_record aleg_uuid /path/to/audio.file)<br><br>A C-Party listen using eavesdrop (originate 2xxx@example.domain&eavesdrop<<a href="mailto:2xxx@example.domain&eavesdrop">mailto:2xxx@example.domain&eavesdrop</a>> aleg_uuid)<br><br><br><br><br><br>-----Original Message-----<br><br>From: freeswitch-users-bounces@lists.freeswitch.org<<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>> [<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of x.liu<br><br>Sent: Monday, February 27, 2012 2:52 PM<br><br>To: freeswitch-users@lists.freeswitch.org<<a href="mailto:freeswitch-users@lists.freeswitch.org">mailto:freeswitch-users@lists.freeswitch.org</a>><br><br>Subject: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br><br><br>Hi Peter,<br><br><br><br>Thanks for your reply!<br><br><br><br>You are right, that is what I have been experiencing but not I wanted.<br><br><br><br>Actually my goal is simply to record the user call by my app (js or java<br><br>via ESL) and at the same time<br><br>I want to hear what the user is saying through my computer's speaker.<br><br><br><br>The recording app seems to be ok ( I tried recording via dialplan<br><br>commands as well as ESL Java app).<br><br>What I am doing now is to bridge the call to my app for recording and to<br><br>a softphone for hearing user's voice.<br><br><br><br>I thought it would work as the wiki says the bridge can bridge the call<br><br>to multiple endpoints at the same time,<br><br>but unfortunately it won't work for me.<br><br><br><br>Any suggestions to achieve my goal please: recording and hearing user's<br><br>voice?<br><br> (Note: my app will do the recording and also allow me send TTS to the<br><br>user)<br><br><br><br>Best,<br><br>Xing<br><br><br><br><br><br><br><br>On 02/27/2012 06:33 PM, Peter Olsson wrote:<br><br>Can you explain what you're trying to achieve by doing this?<br><br><br><br>If you bridge to multiple destinations, the first destination that answers will create the actual call - all other legs are terminated.<br><br><br><br>/Peter<br><br><br><br>Från: freeswitch-users-bounces@lists.freeswitch.org<<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>> [freeswitch-users-bounces@lists.freeswitch.org<<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">mailto:freeswitch-users-bounces@lists.freeswitch.org</a>>] för x.liu [x.liu@hw.ac.uk<<a href="mailto:x.liu@hw.ac.uk">mailto:x.liu@hw.ac.uk</a>>]<br><br>Skickat: den 27 februari 2012 19:11<br><br>Till: freeswitch-users@lists.freeswitch.org<<a href="mailto:freeswitch-users@lists.freeswitch.org">mailto:freeswitch-users@lists.freeswitch.org</a>><br><br>Ämne: Re: [Freeswitch-users] How to bridge a call to an extension defined in dialplan<br><br><br><br>Hi again,<br><br><br><br>It looks like there is something wrong relating to the [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL],<br><br>but no idea how this happened.<br><br><br><br>A bit desperately to solve this problem ASAP as we will have a demo in a few days.<br><br><br><br>Please could anybody give any clue?<br><br><br><br>Many thanks!<br><br><br><br>On 02/27/2012 11:34 AM, x.liu wrote:<br><br>Now I tried it at a different machine with different version of FS.<br><br>The behaviour is slightly different. This time the softphone is ringing but the recording extension is not correctly executed.<br><br>The version for this test is "FreeSWITCH Version 1.0.head (git-e566057 2011-12-14 10-27-31 -0500)"<br><br> (The FS version for previous tries is "FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600)" )<br><br><br><br>I copy part of terminal message here, hope it is not too much message for the email list.<br><br><br><br>--------------------------------------------------<br><br>09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>> [7596ad60-6133-11e1-9141-8b50a5b8b13b]<br><br>09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->74997 in context public<br><br>09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/1003@x.x.x.x<<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>> [75976d18-6133-11e1-9145-8b50a5b8b13b]<br><br>09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/74999@x.x.x.x<<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>> [75977f24-6133-11e1-914a-8b50a5b8b13b]<br><br>09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [7597aab2-6133-11e1-914e-8b50a5b8b13b]<br><br>09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context public<br><br>09:04.180223 [NOTICE] switch_channel.c:920 New Channel sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [75980192-6133-11e1-9150-8b50a5b8b13b]<br><br>09:04.180223 [NOTICE] switch_ivr.c:1711 Transfer sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>> to XML[1003@public]<br><br>09:04.180223 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1003@x.x.x.x<<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]<br><br>09:04.180223 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/74999@x.x.x.x<<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]<br><br>09:04.180223 [INFO] mod_dptools.c:2897 Originate Failed. Cause: ORIGINATOR_CANCEL<br><br>09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context public<br><br>09:04.180223 [NOTICE] switch_core_session.c:1397 Session 2 (sofia/internal/1003@x.x.x.x<<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>>) Ended<br><br>09:04.180223 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/1003@x.x.x.x<<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>><<a href="mailto:sofia/internal/1003@x.x.x.x">mailto:sofia/internal/1003@x.x.x.x</a>> [CS_DESTROY]<br><br>09:04.180223 [NOTICE] switch_ivr.c:1711 Transfer sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>> to XML[1003@default]<br><br>09:04.180223 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->1003 in context default<br><br>09:04.180223 [NOTICE] sofia.c:427 Hangup sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [CS_EXECUTE] [BLIND_TRANSFER]<br><br>09:04.201188 [NOTICE] switch_core_session.c:1397 Session 4 (sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>>) Ended<br><br>09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [CS_DESTROY]<br><br>09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *1 execute_extension::dx XML features<br><br>09:04.201188 [INFO] mod_dialplan_xml.c:481 Processing anonymous<anonymous>->74999 in context public<br><br>09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *2 record_session:: recordings/anonymous.2012-02-27-11-09-04.wav<br><br>09:04.201188 [NOTICE] switch_core_session.c:1397 Session 3 (sofia/internal/74999@x.x.x.x<<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>>) Ended<br><br>09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/74999@x.x.x.x<<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>><<a href="mailto:sofia/internal/74999@x.x.x.x">mailto:sofia/internal/74999@x.x.x.x</a>> [CS_DESTROY]<br><br>09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *3 execute_extension::cf XML features<br><br>09:04.201188 [NOTICE] sofia.c:6134 Hangup sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [CS_EXECUTE] [ORIGINATOR_CANCEL]<br><br>09:04.201188 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *4 execute_extension::att_xfer XML features<br><br>09:04.201188 [NOTICE] switch_core_session.c:1397 Session 5 (sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>>) Ended<br><br>09:04.201188 [NOTICE] switch_core_session.c:1399 Close Channel sofia/internal/anonymous@x.x.x.x<<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>><<a href="mailto:sofia/internal/anonymous@x.x.x.x">mailto:sofia/internal/anonymous@x.x.x.x</a>> [CS_DESTROY]<br><br>09:04.201188 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:1003@y.y.y.y:5062 [759a6630-6133-11e1-9158-8b50a5b8b13b]<br><br>09:04.740223 [NOTICE] sofia.c:5462 Ring-Ready sofia/internal/sip:1003@y.y.y.y:5062!<br><br>09:04.740223 [NOTICE] mod_sofia.c:2470 Ring-Ready sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>>!<br><br>09:04.740223 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>>!<br><br>09:07.700221 [NOTICE] sofia.c:6040 Channel [sofia/internal/sip:1003@y.y.y.y:5062] has been answered<br><br>09:07.700221 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/external/anonymous@sipgate.co.uk<<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>><<a href="mailto:sofia/external/anonymous@sipgate.co.uk">mailto:sofia/external/anonymous@sipgate.co.uk</a>>!<br><br>-------------------------------------------------<br><br><br><br><br><br>Hi,<br><br><br><br>H tried this way<br><br><br><br><action application="bridge" data="sofia/internal/1003%${sip_profile},sofia/internal/74991@${domain}"/><br><br><br><br>74991 is reached and I heard the prompt. The 1003 softphone is terminated immediately after it is reached.<br><br>then extension 74991 hung up itself as well.<br><br><br><br>I saw the terminal message:<br><br>mod_dptools.c:2897 Originate Failed. Cause: ORIGINATOR_CANCEL<br><br><br><br>What would be the reason for that?<br><br><br><br>Thanks,<br><br>Xing<br><br><br><br><br><br><br><br>On 02/26/2012 10:29 PM, Liu, Xingkun wrote:<br><br><br><br><br><br>Hello,<br><br><br><br>I'd like to bridge an incoming to two endpoints simultaneuously:<br><br>one is a softphone which is registered to FS (to user 1003),<br><br>another is just an extension which is defined in conf/dialplan/public.xml<br><br><br><br>It looks like:<br><br><br><br><extension name="testRecording_bridge"><br><br> <condition field="destination_number" expression="^(74990)$"><br><br> <action application="bridge" data="sofia/internal/1003%${sip_profile},sofia/internal/74991%${sip_profile}"/><br><br> </condition><br><br></extension><br><br><br><br><extension name="testRecordingOnly"><br><br> <condition field="destination_number" expression="^74991$"><br><br> <action application="set" data="record_waste_resources=true"/><br><br> <action application="javascript" data="testRecordFile.js"/><br><br> </condition><br><br> </extension><br><br><br><br>The incoming call dial to extension 74990, from there I want to bridge the call to the sofphone at 1003 and<br><br>to the extension 74991.<br><br><br><br>The problem is that FS can not locate user 74991. So my question is what else I need to modify in order<br><br>to bridge the call also to extension 74991 at the same time?<br><br><br><br>I did lots of Googling, checked the wiki and tried various ways but still without success.<br><br><br><br>Any advices please?<br><br><br><br>Many Thanks!<br><br>Xing<br><br><br><br>________________________________<br><br><br><br>Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.<br><br><br><br>Heriot-Watt University is a Scottish charity registered under charity number SC000278.<br><br><br><br><br><br>_________________________________________________________________________<br><br>Professional FreeSWITCH Consulting Services:<br><br>consulting@freeswitch.org<<a href="mailto:consulting@freeswitch.org">mailto:consulting@freeswitch.org</a>><<a href="mailto:consulting@freeswitch.org">mailto:consulting@freeswitch.org</a>><<a href="mailto:consulting@freeswitch.org">mailto:consulting@freeswitch.org</a>><br><br><a 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href="http://www.freeswitch.org">http://www.freeswitch.org</a><br><br><br><br><br><br><br><br>--<br><br>Heriot-Watt University is a Scottish charity<br><br>registered under charity number SC000278.<br><br><br><br>Heriot-Watt University is the Sunday Times<br><br>Scottish University of the Year 2011-2012<br><br><br><br><br><br><br><br>_________________________________________________________________________<br><br>Professional FreeSWITCH Consulting Services:<br><br>consulting@freeswitch.org<<a href="mailto:consulting@freeswitch.org">mailto:consulting@freeswitch.org</a>><br><br><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a><br><br><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><br><a href="http://www.cudatel.com">http://www.cudatel.com</a><br><br><br><br>Official FreeSWITCH Sites<br><br><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><br><br><a 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href="http://www.freeswitch.org">http://www.freeswitch.org</a><br><br>________________________________<br><br>Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.<br><br>Heriot-Watt University is a Scottish charity registered under charity number SC000278.<br><br>________________________________<br><br>Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.<br><br>Heriot-Watt University is a Scottish charity registered under charity number SC000278.</span><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><div class=MsoNormal align=center style='text-align:center'><hr size=2 width="100%" align=center></div><p class=MsoNormal><br><b><span style='font-family:"Arial","sans-serif"'>Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012.</span></b><span style='font-family:"Arial","sans-serif"'> </span><br><br><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Heriot-Watt University is a Scottish charity registered under charity number SC000278. </span><o:p></o:p></p></div></body></html>