<div dir="ltr">hi,<br><br>can someone tell me what is happening here. I figured that freeswitch is not leaving the extension,<br>it just rexecutes it. I would like it do another action when there is a value inside pre_transfer_caller_id_number<br><br>[INFO] switch_channel.c:3062 sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a> Flipping CID from "" <18> to "Outbound Call" <14><br>[DEBUG] switch_core_state_machine.c:216 (sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a>) State Change CS_ROUTING -> CS_EXECUTE<br>[DEBUG] switch_core_session.c:1388 Send signal sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a> [BREAK]<br>[DEBUG] switch_core_state_machine.c:528 (sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a>) State ROUTING going to sleep<br>[DEBUG] switch_core_state_machine.c:472 (sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a>) Running State Change CS_EXECUTE<br>[DEBUG] switch_core_state_machine.c:535 (sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a>) State EXECUTE<br>[DEBUG] mod_sofia.c:178 sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a> SOFIA EXECUTE<br>[DEBUG] switch_core_state_machine.c:258 sofia/internal/<a href="http://sip:14@10.0.200.14:5060">sip:14@10.0.200.14:5060</a> Standard EXECUTE<br><br><br> <extension name="Extension"><br> <condition field="destination_number" expression="^([0-9][0-9])$"><br> <action application="export" data="dialed_extension=$1"/><br> <action application="set" data="call_timeout=90"/><br> <action application="set" data="continue_on_fail=true"/><br> <action application="set" data="hangup_after_bridge=true"/><br><br> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/><br> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/><br> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/><br><br> <action application="bridge" data="[leg_timeout=20]user/${dialed_extension}"/><br><br> </condition><br> </extension><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Mar 10, 2015 at 4:57 PM, Paul Atreides <span dir="ltr"><<a href="mailto:paul.atreides83@googlemail.com" target="_blank">paul.atreides83@googlemail.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"><div><div><span class=""><div>>In other words, if user A did a blind x-fer of caller C to user B and
user B doesn't answer (for whatever failure reason) then >caller C would
start ringing back to user A? Just making sure we understand the scope
of the feature you're implementing.<br><br></div></span>Exactly, at the moment the GXP disconnects A and makes a new call to B by using the default dial plan. So freeswitch thinks its a normal call and precedes as usual. I need a condition where I can check if its a transfer / attended call or not<br><br></div><div>In the log below I found the channel variable variable_pre_transfer_caller_id_number and variable_sip_refer_to but these are are only set when it is a blind transfer. But for some reason there is no match for the condition. <br></div><div><br></div>Here is my default dial plan<br><a href="http://pastebin.freeswitch.org/23993" target="_blank">http://pastebin.freeswitch.org/23993</a><br><br></div>Here is the debug from the console when I start the transfer.<br><a href="http://pastebin.freeswitch.org/23994" target="_blank">http://pastebin.freeswitch.org/23994</a><br><br><br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Mar 9, 2015 at 7:18 PM, 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:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote"><span>On Sat, Mar 7, 2015 at 9:03 AM, Paul Atreides <span dir="ltr"><<a href="mailto:paul.atreides83@googlemail.com" target="_blank">paul.atreides83@googlemail.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"><div><div><div><div>When I do a blind transfer then I want freeswitch to call back the origin who initiated the call.<br></div>But I am not able the capture the transfer event?<br></div></div></div></div></blockquote><div><br></div></span><div>In other words, if user A did a blind x-fer of caller C to user B and user B doesn't answer (for whatever failure reason) then caller C would start ringing back to user A? Just making sure we understand the scope of the feature you're implementing.<br><br></div><div>How does the GXP do the x-fer? Some kind of hook-flash and DTMF code? Can you pastebin the dialplan that the transferor uses when sending the call? <br><br></div><div>-MC<br><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span><div dir="ltr"><div><div><div><br></div><code><</code><code>condition</code> <code>field</code><code>=</code><code>"destination_number"</code> <code>expression</code><code>=</code><code>"^att_xfer$"</code><code>><br></code><code><</code><code>condition</code> <code>field</code><code>=</code><code>"destination_number"</code> <code>expression</code><code>=</code><code>"^is_transfer$"</code><code>><br><br></code></div>They seem do be ignored by the dialplan. Is there a list what kind of values destionation_number can have besides the called numbers?<br><br><br></div>I am doing the transfer with a grandstream gxp2140<br></div>
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