<div dir="ltr"><div><div>My conference end logic is - when the moderator ends the call, all the conference member gets kicked out, that's why we see this log for every number </div><div> </div><div>: <span style="white-space:pre-wrap">2016-09-09 10:54:19.892289 [NOTICE] switch_core_state_machine.c:</span><span style="white-space:pre-wrap">31<wbr>5 sofia/external/+918558897592 has executed the last dialplan instruction, hanging up.</span></div></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">Moderator Number - 9716517818</span></div><div><br></div><div><br></div><div>Conference Start by the moderator at : 2016-09-09 10:38:42.752327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/+919716517818 [9402c95e-7679-11e6-99e4-<wbr>a53a7934abe1]</div><div><br></div><div><br></div><div>Conferece End by the moderator: 2016-09-09 10:54:19.912289 [NOTICE] switch_core_state_machine.c:<wbr>317 Hangup sofia/external/+919716517818 [CS_EXECUTE] [NORMAL_CLEARING]</div><div><br></div><div><br></div><div>In-between the drop of a conference member happens at 206-09-09 10:46:39.232313 [NOTICE] sofia.c:952 Hangup sofia/external/+918558897588 [CS_EXECUTE] [NORMAL_CLEARING]</div><div><br></div><div>After this my Program retries to connect to it several time and ends at: 2016-09-09 10:47:35.072286 [NOTICE] sofia.c:7539 Hangup sofia/external/918558897588 [CS_EXECUTE] [NO_USER_RESPONSE]</div><div><br></div><div>You may only look at the logs of the hangup of :</div><div><br></div><div><a href="http://pastebin.com/gFgELQnv">http://pastebin.com/gFgELQnv</a><br></div><div><br></div><div><br></div><div><div>I am using only Freeswitch which interacts with my GSM gateway, I have no clue from where the Asterisk context is coming. I am blind to the logic behind gateway.</div></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Sep 10, 2016 at 11:42 AM, Dave Horton <span dir="ltr"><<a href="mailto:daveh@beachdognet.com" target="_blank">daveh@beachdognet.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">For number <span style="color:rgb(51,51,51);font-family:consolas,menlo,monaco,'lucida console','liberation mono','dejavu sans mono','bitstream vera sans mono',monospace,serif">918558897592 I only see you (your freeswitch dialplan app) hanging up the call:</span><div><br><pre style="word-wrap:break-word;white-space:pre-wrap">2016-09-09 10:54:19.892289 [NOTICE] switch_core_state_machine.c:<wbr>315 sofia/external/+918558897592 has executed the last dialplan instruction, hanging up.<br></pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">So I guess I would look at the dialplan.</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">There is also a lot of retransmissions between your freeswitch and your asterisk gateway, which is maybe not related to your problem but it doesn’t appear to me like you have a quality network situation.</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">The log is a bit difficult to read, because it doesn’t seem to contain a full example of an INVITE - OK and then call drop.</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">Your asterisk is also requiring session timers (with a 15 min expiry) so another thing to validate is that those session refreshes are actually being sent.</pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap">Can you get a simpler log showing an incoming call from setup until it drops?</pre><div><div class="h5"><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><pre style="word-wrap:break-word;white-space:pre-wrap"><br></pre><div><div>On Sep 10, 2016, at 1:56 AM, Deepika Yadav <<a href="mailto:deepikay@iiitd.ac.in" target="_blank">deepikay@iiitd.ac.in</a>> wrote:</div><br><div><div dir="ltr"><div>I haved pasted a section of SIP trace logs here :<br></div><div><br></div><div><a href="http://pastebin.com/5CUpY63c" target="_blank">http://pastebin.com/5CUpY63c<br></a></div><div><br></div><div>Freeswitch - CALL REJECTED logs are coming for the call drops of these phone numbers - <span style="color:rgb(51,51,51);font-family:consolas,menlo,monaco,"lucida console","liberation mono","dejavu sans mono","bitstream vera sans mono",monospace,serif;line-height:24px">918558897586, </span><span style="color:rgb(51,51,51);font-family:consolas,menlo,monaco,"lucida console","liberation mono","dejavu sans mono","bitstream vera sans mono",monospace,serif;line-height:24px">918558897592.</span><br></div><div><br></div><div><span style="background-color:rgb(255,255,255)"><span style="color:rgb(51,51,51);font-family:consolas,menlo,monaco,"lucida console","liberation mono","dejavu sans mono","bitstream vera sans mono",monospace,serif;line-height:24px"><br></span></span></div><div>Regards,</div><div>Deepika</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 7, 2016 at 5:25 PM, Benjamin Cropley <span dir="ltr"><<a href="mailto:benjamin.cropley@gmail.com" target="_blank">benjamin.cropley@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">I would look at a SIP trace first and try to better understand how the calls are dropping<div><br></div><div>eg..</div><div><br></div><div>1. Does the caller suddenly stop sending audio, but no SIP packet to indicate the end of the session. Therefore FreeSWITCH would time out (on receipt of no audio) and terminate. </div><div>2. Does the caller or FreeSWITCH send a termination (in the form of a BYE) to the other party? If this is the case, look at the logs for something that provoked it.</div><div><br></div><div>Good luck,</div><div>Ben</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Mon, Sep 5, 2016 at 11:48 AM, Deepika Yadav <span dir="ltr"><<a href="mailto:deepikay@iiitd.ac.in" target="_blank">deepikay@iiitd.ac.in</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Hi ,<div><br></div><div>I am using conference call service of Freeswitch over ESL with 21- 22 members over GSM for around 1 hour. One problem that I am facing is frequent call drops of the connected callers.</div><div><br></div><div>My program has a logic of redialing a caller in case of disconnection, if the caller is not reachable due to some reason the redial happens three times. I am maintaining sufficient balance to keep 22 channels in parallel and also for the duration of calls.</div><div><br></div><div>One thing that I am observing which I am not sure to be the cause is when I increased the redial trial to five times, the frequency of call drops increased.</div><div><br></div><div>Is the dialing attempt count should be taken care for the balance maintenance or there is some other cause ?</div><div><br></div><div>The Freeswitch logs are of type - CALL REJECTED, USER BUSY</div><div><br></div><div>Regards,</div><div>Deepika</div></div>
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