JM,<br><br>That actually explains a lot. I appreciate the information. I can definitely see where the relative expense of a hard phone would make this kind of application quite valuable. I hope you guys are able to iron out the details. Also, maybe you could see if Mitch Capper could add line appearances to FSClient. :)<br>
<br>-MC<br><br><div class="gmail_quote">On Mon, Sep 17, 2012 at 3:42 PM, <a href="mailto:Jmesquita@freeswitch.org">Jmesquita@freeswitch.org</a> <span dir="ltr"><<a href="mailto:jmesquita@freeswitch.org" target="_blank">jmesquita@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor="#FFFFFF"><div>MC, I believe I have to make a statement here in favor of some realities that are different from the ones most people on this list live in. I live in South America and an IP Phone here does not get lower than 120usd per unit. And I am talking about the cheapest yea link phone model. In brazil, where I come from, this is even higher.</div>
<div><br></div><div>Taking that into consideration, several lower end ip phones started to appear as well as hybrid systems where the majority of the extensions are still analog. In a system like that, CTI applications like the one our friend is describing is really the way out to really add value to a solution.</div>
<div><br></div><div>Even on the asterisk world we see applications like these. For example the flash operator panel and the HUD for asterisk. I believe most of us have heard of it before.</div><div><br></div><div>Anyhow, I didn't mean to write an essay on the subject but I see this kind of feature being constantly rejected by the community and I really can't understand why that is.</div>
<div><br>João Mesquita</div><div><br>On 17/09/2012, at 01:39 p.m., Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br><br></div><div></div><blockquote type="cite">
<div>Or get a hard phone that has a hold button and at least two line keys.<br>-MC<br><br><div class="gmail_quote">On Mon, Sep 17, 2012 at 5:19 AM, João Mesquita <span dir="ltr"><<a href="mailto:jmesquita@freeswitch.org" target="_blank">jmesquita@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p>From personal experience, I believe that how you described is the right way to do it. The only caveat is that you will have to add variables do the channels so you can properly track what is going on in the cdrs . If you don't process cdrs, then it is all good. Careful with pickup and such too...</p>
<div><div>
<div class="gmail_quote">On Sep 17, 2012 3:53 AM, "Alexander Haugg" <<a href="mailto:Alexander.Haugg@c4b.de" target="_blank">Alexander.Haugg@c4b.de</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div link="blue" vlink="purple" lang="DE">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi MC,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">thank</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">you for the
</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">answer</span><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">To your question, all call legs in this scenario (outgoing or incoming direction) are connected over a sip trunk of a pbx.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Scenario:<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">My CTI<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> | A<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> | | event socket<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> V |<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Freeswitch Sip Trunk<u></u><u></u></span></p>
<p class="MsoNormal" style="margin-left:35.1pt;text-indent:35.4pt"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><- PBX<u></u><u></u></span></p>
<p class="MsoNormal" style="margin-left:35.1pt;text-indent:35.4pt"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">->
<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">My Client control (CTI) the call legs over the event socket interface and the call legs are only legs over the sip trunk to or from the PBX.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">After some tests i have found a possible solution:<u></u><u></u></span></p>
<p><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><span>-<span style="font:7.0pt "Times New Roman"">
</span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Leg A and Leg B are bridged (all legs get the flag park_after_bridge = true)<u></u><u></u></span></p>
<p><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><span>-<span style="font:7.0pt "Times New Roman"">
</span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">For Consultation i park Leg B and transfer this Leg to Moh in my default context of my dialplan<u></u><u></u></span></p>
<p><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><span>-<span style="font:7.0pt "Times New Roman"">
</span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">i originate Leg C and bridge this Leg with Leg A (Leg C have the flag park_after_bridge = true too) Consultation is comlete now.<u></u><u></u></span></p>
<p><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><span>-<span style="font:7.0pt "Times New Roman"">
</span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">with the principle same think i can toggle Leg A <-> Leg B and Leg A <-> Leg C.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">What is your think for this solution? I have tested this on the FS CLI and it works.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Von:</span></b><span style="font-size:10.0pt;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>Im Auftrag von </b>Michael Collins<br>
<b>Gesendet:</b> Freitag, 14. September 2012 19:13<br>
<b>An:</b> FreeSWITCH Users Help<br>
<b>Betreff:</b> Re: [Freeswitch-users] Consultation Call via event_socket interface<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal" style="margin-bottom:12.0pt">Hi Alex,<br>
<br>
Welcome to the FreeSWITCH mail list! <br>
<br>
First question for you: what kind of telephone are you using? The reason I ask is that this kind of function is trivially achieved with a good hard phone, like a Polycom, Yealink, Snom, or Cisco with two or more line keys. If you can use a hard phone w/ multiple
line keys then you don't even need to mess with the dialplan, uuid_bridge, etc.<br>
<br>
-MC<u></u><u></u></p>
<div>
<p class="MsoNormal">On Fri, Sep 14, 2012 at 12:24 AM, Alexander Haugg <<a href="mailto:Alexander.Haugg@c4b.de" target="_blank">Alexander.Haugg@c4b.de</a>> wrote:<u></u><u></u></p>
<div>
<div>
<p class="MsoNormal">Hi All,<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">I’m new on the mailing list.<u></u><u></u></p>
<p class="MsoNormal">I have a problem with a call scenario.<u></u><u></u></p>
<p>-<span style="font-size:7.0pt"> </span>Channel A and channel B are bridged (A is my own channel and B is my calling partner)<u></u><u></u></p>
<p>-<span style="font-size:7.0pt"> </span>Now i set channel B on hold with the command „uuid_hold xxx“ and create a new channel to C with the command:<u></u><u></u></p>
<p>bgapi originate {channel_csid=num,accountcode=num,origination_caller_id_name='num@ip',origination_caller_id_number=num@ip}sofia/external/num@ip &park<u></u><u></u></p>
<p>this works correctly, the partner C answer and the channel is established.<u></u><u></u></p>
<p>-<span style="font-size:7.0pt"> </span>Now the Problem:<u></u><u></u></p>
<p>I try to bridge the channel a to channel c with the uuid_bridge command, now the channel b will hangup, why? The variable hangup_after_bridge is by default false.<u></u><u></u></p>
<p>Other problem: channel A can hear the voice of channel C but not speak with him, channel C can hear and speak. But this problem is not the important think at the moment.<u></u><u></u></p>
<p class="MsoNormal">Is there a general problem in my plan to do that?<u></u><u></u></p>
<p class="MsoNormal">Is it a better plan to do this over the dialplan?<u></u><u></u></p>
<p class="MsoNormal">The next step in this scenario is to toggle the connection A -> B and A -> C.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Thanks for your help!<u></u><u></u></p>
<p class="MsoNormal">Nice regards,<u></u><u></u></p>
<p class="MsoNormal">Alex<u></u><u></u></p>
</div>
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