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</o:shapelayout></xml><![endif]--></head><body lang=SV link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='mso-fareast-language:EN-US'>Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred via data connection to the asterisk server.<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>Seems theres a problem with the trunk then.<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>What does ”sip show registry” tell you?<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>(asterisk -r in console and then sip show registry)<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>It should show a status of ”Registred” to your trunk operator.<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b>Från:</b> Ivan Demkovitch <idemkovitch@yahoo.com> <br><b>Skickat:</b> den 15 november 2018 18:01<br><b>Till:</b> Sebastian Nielsen <sebastian@sebbe.eu>; 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com><br><b>Ämne:</b> Re: SV: [asterisk-users] Queue not dialing out to cell phone for some reason<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>Sebastian,<o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10711"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><o:p> </o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10713"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>I don't think it has to do anything with registration. It is dialing through the SIP trunk, so it goes out as normal cell phone call.<o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10715"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>Also, why I don't see anything in a log? I see only first 2 members being dialed. <o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10692"><p class=MsoNormal style='margin-bottom:12.0pt;background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><o:p> </o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10698"><div id="yui_3_16_0_ym19_1_1542243121702_10697"><div id="yui_3_16_0_ym19_1_1542243121702_10696"><div id="yui_3_16_0_ym19_1_1542243121702_10695"><div class=MsoNormal align=center style='text-align:center;background:white'><span style='font-size:10.0pt;font-family:"Arial",sans-serif;color:black'><hr size=1 width="100%" align=center></span></div><p class=MsoNormal style='background:white'><b><span style='font-size:10.0pt;font-family:"Arial",sans-serif;color:black'>From:</span></b><span style='font-size:10.0pt;font-family:"Arial",sans-serif;color:black'> Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>><br><b>To:</b> 'Ivan Demkovitch' <<a href="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>>; 'Asterisk Users Mailing List - Non-Commercial Discussion' <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> <br><b>Sent:</b> Thursday, November 15, 2018 10:58 AM<br><b>Subject:</b> SV: [asterisk-users] Queue not dialing out to cell phone for some reason</span><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10721"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><o:p> </o:p></span></p><div id=yiv7898733751><div id="yui_3_16_0_ym19_1_1542243121702_10720"><div id="yui_3_16_0_ym19_1_1542243121702_10719"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>I would suspect that the cell phone does use battery saving causing the SIP application to lose registration with the server. Would also suggest using TCP with a fairly short keepalive to prevent the cellular network from tearing down the connection to the asterisk server.<o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10728"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>You need to go into android settings and make sure the SIP client is whitelisted in battery management.<o:p></o:p></span></p></div><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div><div id=yiv7898733751yqt97475><div id="yui_3_16_0_ym19_1_1542243121702_10732"><div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;border-color:currentColor currentColor;border-image: none' id="yui_3_16_0_ym19_1_1542243121702_10731"><div id="yui_3_16_0_ym19_1_1542243121702_10730"><p class=MsoNormal style='background:white'><b><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>Från:</span></b><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> <b>För </b>Ivan Demkovitch<br><b id="yui_3_16_0_ym19_1_1542243121702_10729">Skickat:</b> den 15 november 2018 17:55<br><b>Till:</b> <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br><b>Ämne:</b> [asterisk-users] Queue not dialing out to cell phone for some reason<o:p></o:p></span></p></div></div></div><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div><div id="yui_3_16_0_ym19_1_1542243121702_10718"><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9283"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>Hello,<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9122"><div id="yui_3_16_0_ym19_1_1542243121702_10733"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9121"><div id="yui_3_16_0_ym19_1_1542243121702_10717"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>I have queues.conf setup with a group like so:<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9145"><div id="yui_3_16_0_ym19_1_1542243121702_10734"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9170"><div id="yui_3_16_0_ym19_1_1542243121702_10722"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>[Sales](StandardQueue)<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9138">announce = first<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9139">member => SIP/FF4C119EEBF8-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9140">member => SIP/FF9EF375CCFC-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9141">member => SIP/13145555555@callcentric ;Eric's cell<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9142">member => SIP/FF1565AABB2D-SLS ;Eric's Yealink<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9169"><div id="yui_3_16_0_ym19_1_1542243121702_10735"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9282"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9357"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>I did trace a call and this is what I see. Only 2 phones (internal) called. External SIP@callcentric is not being called.<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9355"><div id="yui_3_16_0_ym19_1_1542243121702_10724"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9356"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'>Any idea why it's not being called?<o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9275"><div><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'> <o:p></o:p></span></p></div></div><div id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9263"><div id="yui_3_16_0_ym19_1_1542243121702_10725"><p class=MsoNormal style='background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><br> -- Executing [1@automated_attendant_normal:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9222"> Caller "aa" <15555555555> has entered the sales queue<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9223"> -- Executing [1@automated_attendant_normal:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9224"> -- Goto (queues,7001,1)<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9225"> -- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9226"> == "aa" <15555555555> entering sales queue<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9227"> -- Executing [7001@queues:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9228"> -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9229"> -- Executing [7001@queues:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9230"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9231"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9232"> -- Called SIP/FF9EF375CCFC-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9233"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9234"> -- Called SIP/FF4C119EEBF8-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9235"> -- SIP/FF4C119EEBF8-SLS-00000437 is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9236"> -- SIP/FF9EF375CCFC-SLS-00000436 is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9237"> -- Nobody picked up in 30000 ms<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9238"> -- Nobody picked up in 30000 ms<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9239"> -- Stopped music on hold on SIP/callcentric15-00000435<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9240"> -- Playing periodic announcement<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9241"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9242"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9243"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9244"> -- Called SIP/FF9EF375CCFC-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9245"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9246"> -- Called SIP/FF4C119EEBF8-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9247"> -- SIP/FF4C119EEBF8-SLS-00000439 is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9248"> -- SIP/FF9EF375CCFC-SLS-00000438 is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9249"> -- Nobody picked up in 30000 ms<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9250"> -- Nobody picked up in 30000 ms<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9251"> -- Stopped music on hold on SIP/callcentric15-00000435<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9252"> -- Playing periodic announcement<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9253"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9254"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9255"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9256"> -- Called SIP/FF9EF375CCFC-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9257"> == Using SIP RTP CoS mark 5<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9258"> -- Called SIP/FF4C119EEBF8-SLS<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9259"> -- SIP/FF4C119EEBF8-SLS-0000043b is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9260"> -- SIP/FF9EF375CCFC-SLS-0000043a is ringing<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9261"> -- Stopped music on hold on SIP/callcentric15-00000435<br id="yiv7898733751yui_3_16_0_ym19_1_1542243121702_9262"> == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'<o:p></o:p></span></p></div></div></div></div></div></div></div><p class=MsoNormal style='margin-bottom:12.0pt;background:white'><span style='font-size:12.0pt;font-family:"Helvetica",sans-serif;color:black'><o:p> </o:p></span></p></div></div></div></div></div></div></body></html>