<div dir="ltr">Hi,<div><br></div><div>If you check following log , There is a almost delay of 7 sec. Please let me know what may be wrong. I have manually hide my client IP and phone number in logs.</div><div><br></div><div>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_ivr_originate.c:3607 Originate Resulted in Success: [sofia/sipinterface_1/<a href="mailto:xxxxxxxxxxx@sbc.voxbeam.com">xxxxxxxxxxx@sbc.voxbeam.com</a>]</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] sofia.c:6858 Channel sofia/sipinterface_1/nuz54@XXX.82.43.202:5060 entering state [early][183]</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_core_media.c:9483 sofia/sipinterface_1/<a href="mailto:xxxxxxxxxxx@sbc.voxbeam.com">xxxxxxxxxxx@sbc.voxbeam.com</a> PAUSE Jitterbuffer</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_core_media.c:9483 sofia/sipinterface_1/nul554@xxx.xx.43.202:5060 PAUSE Jitterbuffer</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] kazoo_event_stream.c:245 increased event stream buffer size to 4849</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_ivr_bridge.c:1594 (sofia/sipinterface_1/<a href="mailto:xxxxxxxxxxx@sbc.voxbeam.com">xxxxxxxxxxx@sbc.voxbeam.com</a>) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_core_state_machine.c:543 (sofia/sipinterface_1/<a href="mailto:xxxxxxxxxxxxx@sbc.voxbeam.com">xxxxxxxxxxxxx@sbc.voxbeam.com</a>) Running State Change CS_EXCHANGE_MEDIA</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] switch_core_state_machine.c:612 (sofia/sipinterface_1/<a href="mailto:xxxxxxxxxx@sbc.voxbeam.com">xxxxxxxxxx@sbc.voxbeam.com</a>) State EXCHANGE_MEDIA</span></p>
<p class=""><span class="">2016-05-08 06:18:30.288192 [DEBUG] mod_sofia.c:628 SOFIA EXCHANGE_MEDIA</span></p>
<p class=""><span class="">2016-05-08 06:18:30.308197 [DEBUG] switch_rtp.c:6694 Correct audio ip/port confirmed.</span></p>
<p class=""><span class=""><b>2016-05-08 06:18:<span style="background-color:rgb(0,255,0)">30</span>.728192 [DEBUG] switch_rtp.c:6694 Correct audio ip/port confirmed.</b></span></p>
<p class=""><span class=""><b>2016-05-08 06:18:<span style="background-color:rgb(255,0,0)">37.</span>168193 [NOTICE] switch_channel.c:1104 New Channel sofia/sipinterface_1/<a href="http://51841@173.82.43.202:5060">51841@173.82.43.202:5060</a> [<a href="mailto:GWbegcciibgkinub3864822@173.82.43.202">GWbegcciibgkinub3864822@173.82.43.202</a>]</b></span></p>
<p class=""><span class="">2016-05-08 06:18:37.168193 [DEBUG] switch_core_state_machine.c:543 (sofia/sipinterface_1/51841@xxx.82.43.202:5060) Running State Change CS_NEW</span></p>
<p class=""><span class="">2016-05-08 06:18:37.168193 [DEBUG] sofia.c:9374 sofia/sipinterface_1/51841@xxx.82.43.202:5060 receiving invite from <a href="http://173.82.43.202:5060">173.82.43.202:5060</a> version: 1.7.0 64bit</span></p>
<p class=""><span class="">2016-05-08 06:18:37.168193 [DEBUG] sofia.c:9486 IP xxx.82.43.202 Approved by acl "trusted[]". Access Granted.</span></p>
<p class=""><span class="">2016-05-08 06:18:37.168193 [DEBUG] sofia.c:6858 Channel sofia/sipinterface_1/51841@xxx.82.43.202:5060 entering state [received][100]</span></p>
<p class=""><span class="">2016-05-08 06:18:37.168193 [DEBUG] sofia.c:6868 Remote SDP:</span></p><p class=""><span class=""><br></span></p><p class=""><span class="">Thanks,</span></p><p class=""><span class="">Abhishake</span></p></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, May 8, 2016 at 1:09 AM, Abhishake Agarwal <span dir="ltr"><<a href="mailto:abhishake.agarwal@gmail.com" target="_blank">abhishake.agarwal@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">Hi,<div><br></div><div>I am using freeswitch 1.6 with Kazoo just for routing calls to external provider</div><div><br></div><div>I added following config in freeswitch.xml</div><div><br></div><div><div><X-PRE-PROCESS cmd="set" data="ignore_early_media=ring_ready"/></div><div><X-PRE-PROCESS cmd="set" data="instant_ringback=true"/></div><div><X-PRE-PROCESS cmd="set" data="return_ring_ready=true"/></div><div><X-PRE-PROCESS cmd="set" data="ringback=%(2000,4000,440.0,480.0)"/></div></div><div><br></div><div>Still there is a delay of 8 sec from dialing to ringback. But when I see logs I see logs instantly when I hit dial button.</div><div><br></div><div>I want something like whatsapp calls, as we press dial button, there should be instant ring.</div><div><br></div><div>Thanks,</div><div>Abhishake</div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, May 4, 2016 at 8:42 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">What you describe will lead to a ring tone followed by a busy signal in some situations. We have had people try this before and every time they do not like the results.<div><br></div><div>Mike<br><div><blockquote type="cite"><div>On May 4, 2016, at 5:13 AM, Abhishake Agarwal <<a href="mailto:abhishake.agarwal@gmail.com" target="_blank">abhishake.agarwal@gmail.com</a>> wrote:</div><br><div><div dir="ltr">Hi,<div><br></div><div>When user dial a number, early media is active only after 7-10 sec. even after early media is active, actual ring comes after 5-10 sec, till that time caller does not hear anything. I am need of feature, so that user can hear at least progress beep sound, till caller does not hear a actual ring from b leg. </div><div><br></div><div>Thanks,</div><div>Abhishake</div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, May 4, 2016 at 10:06 AM, Sergey Safarov <span dir="ltr"><<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">What must do FreeSwitch after 180 ringing or 183 early media messages?</p>
<br><div class="gmail_quote"><div dir="ltr">On Wed, May 4, 2016, 06:41 Sergey Safarov <<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Look at <a href="https://freeswitch.org/jira/browse/FS-9118" target="_blank">FS-9118</a><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">ср, 4 мая 2016 г. в 4:31, Abhishake Agarwal <<a href="mailto:abhishake.agarwal@gmail.com" target="_blank">abhishake.agarwal@gmail.com</a>>:<br></div></div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>I need beep ring until 180 ring is not received or before early media. Please help me resolving this issue.</div></div></blockquote></div></blockquote></div></blockquote></div></div></div></blockquote></div><br></div></div><br>_________________________________________________________________________<br>
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