<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Hi Stefan,<div class="">IMHO NAT would be that problem. Check if you have enabled SIP helpers on router, try it disable, try set static port redirection to phones and set RTP ports in phone configuration (every phone must have different ports range!). Even better, make VPN between Asterisk and phones and leave it in one network without NAT.</div><div class=""><br class=""></div><div class=""><br class=""></div><div class=""><div class=""><span style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; display: inline !important; float: none;" class="">S pozdravem</span><br style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;" class=""><span style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; display: inline !important; float: none;" class="">Tomáš Holý</span><br style="color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;" class=""></div><div class=""><br class="webkit-block-placeholder"></div><div class=""><br class=""></div><div><blockquote type="cite" class=""><div class="">14. 2. 2020 v 9:45, Stefan Viljoen <<a href="mailto:viljoens@verishare.co.za" class="">viljoens@verishare.co.za</a>>:</div><br class="Apple-interchange-newline"><div class=""><div class="">Hi, do you have NAT between Asterisk and agent phones? <br class=""><br class="">S pozdravem<br class="">Tomáš Holý<br class=""><br class="">Hi Tomas<br class=""><br class="">Thanks for replying.<br class=""><br class="">Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud.<br class=""><br class="">A typical sip.conf phone configuration on the remote server for the site is<br class=""><br class="">[general]<br class="">session-timers=refuse<br class="">disallow=all<br class="">allow=g729:20<br class="">allow=ulaw<br class="">allow=alaw<br class="">fromuser=xxx<br class="">useragent=xxx<br class="">callcounter=yes<br class="">alwaysauthreject=yes<br class="">allowguest=no<br class="">jbnable=yes<br class="">jbforce=no<br class="">jbimpl=adaptive<br class="">jblog=no<br class="">jbmaxsize=200<br class="">jbresyncthreshold=1000<br class="">externaddr=xx.xx.xx.xx<br class="">localnet=xx.xx.xx.xx/255.255.255.0<br class=""><br class="">[xxxx]<br class="">type=peer<br class="">user=xxxx<br class="">secret=xxxx<br class="">host=dynamic<br class="">disallow=all<br class="">allow=g729<br class="">allow=ulaw<br class="">allow=alaw<br class="">dtmfmode=rfc2833<br class="">context=xxxx<br class="">call-limit=1<br class="">limitonpeers=yes<br class="">callgroup=1<br class="">pickupgroup=1<br class="">deny=0.0.0.0/0.0.0.0<br class="">permit=xxx.xxx.xxx.xxx/255.255.255.255<br class="">nat=force_rport,comedia<br class=""><br class="">Not sure if that helps much.<br class=""><br class="">Thanks for the reply!<br class=""><br class=""><blockquote type="cite" class="">13. 2. 2020 v 19:06, Stefan Viljoen <<a href="mailto:viljoens@verishare.co.za" class="">viljoens@verishare.co.za</a>>:<br class=""><br class=""><br class="">Hi all<br class=""><br class="">Asterisk 13 instance - I’ve got a situation in an agent queue that an agent will be talking to one person, then suddenly the same agent will be talking to another person who was talking to another agent.<br class=""><br class="">The calls do not switch around between the two agents, the “losing” agent will just suddenly have silence in his handset and the other agent will now be talking to “his” customer.<br class=""><br class="">The original customer is simply cut off instantly the moment this happens.<br class=""><br class="">This happens randomly. I have yet to collect log output and capture the CLI, etc. but anybody ever heard of this happening?<br class=""><br class="">It is as if agent channels get randomly reassigned / lose their audio channel from outside the Asterisk - one channel is disconneted that a customer was on and certain agents are suddenly talking to someone else and lose the original caller they were busy with.<br class=""><br class="">Where can I even being to look?<br class=""><br class="">Thx!<br class=""><image001.png><br class=""></blockquote><br class=""><br class="">-- <br class="">_____________________________________________________________________<br class="">-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" class="">http://www.api-digital.com</a> --<br class=""><br class="">Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" class="">https://community.asterisk.org/</a><br class=""><br class="">New to Asterisk? Start here:<br class=""> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" class="">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br class=""><br class="">asterisk-users mailing list<br class="">To UNSUBSCRIBE or update options visit:<br class=""> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" class="">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></div></blockquote></div><br class=""></div></body></html>