<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">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;" class="">918558897592 I only see you (your freeswitch dialplan app) hanging up the call:</span><div class=""><br class=""><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">2016-09-09 10:54:19.892289 [NOTICE] switch_core_state_machine.c:315 sofia/external/+918558897592 has executed the last dialplan instruction, hanging up.<br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">So I guess I would look at the dialplan.</pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">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="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">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="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">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="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class="">Can you get a simpler log showing an incoming call from setup until it drops?</pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><pre style="font-variant-ligatures: normal; orphans: 2; widows: 2; word-wrap: break-word; white-space: pre-wrap;" class=""><br class=""></pre><div><div class="">On Sep 10, 2016, at 1:56 AM, Deepika Yadav <<a href="mailto:deepikay@iiitd.ac.in" class="">deepikay@iiitd.ac.in</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div class="">I haved pasted a section of SIP trace logs here :<br class=""></div><div class=""><br class=""></div><div class=""><a href="http://pastebin.com/5CUpY63c" class="">http://pastebin.com/5CUpY63c<br class=""></a></div><div class=""><br class=""></div><div class="">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" class="">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" class="">918558897592.</span><br class=""></div><div class=""><br class=""></div><div class=""><span style="background-color:rgb(255,255,255)" class=""><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" class=""><br class=""></span></span></div><div class="">Regards,</div><div class="">Deepika</div></div><div class="gmail_extra"><br class=""><div class="gmail_quote">On Wed, Sep 7, 2016 at 5:25 PM, Benjamin Cropley <span dir="ltr" class=""><<a href="mailto:benjamin.cropley@gmail.com" target="_blank" class="">benjamin.cropley@gmail.com</a>></span> wrote:<br class=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr" class="">I would look at a SIP trace first and try to better understand how the calls are dropping<div class=""><br class=""></div><div class="">eg..</div><div class=""><br class=""></div><div class="">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 class="">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 class=""><br class=""></div><div class="">Good luck,</div><div class="">Ben</div></div><div class="gmail_extra"><br class=""><div class="gmail_quote"><div class=""><div class="h5">On Mon, Sep 5, 2016 at 11:48 AM, Deepika Yadav <span dir="ltr" class=""><<a href="mailto:deepikay@iiitd.ac.in" target="_blank" class="">deepikay@iiitd.ac.in</a>></span> wrote:<br class=""></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=""><div class="h5"><div dir="ltr" class="">Hi ,<div class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class="">Is the dialing attempt count should be taken care for the balance maintenance or there is some other cause ?</div><div class=""><br class=""></div><div class="">The Freeswitch logs are of type - CALL REJECTED, USER BUSY</div><div class=""><br class=""></div><div class="">Regards,</div><div class="">Deepika</div></div>
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