<html><head></head><body><div style="color:#000; background-color:#fff; font-family:Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:16px"><div id="yui_3_16_0_ym19_1_1542243121702_12653"><span>Sebastian,</span></div><div id="yui_3_16_0_ym19_1_1542243121702_12654"><span><br></span></div><div id="yui_3_16_0_ym19_1_1542243121702_12655"><span>Well, this can't be problem with trunk because:</span></div><div id="yui_3_16_0_ym19_1_1542243121702_12656"><span>1. Call coming from outside, so trunk works</span></div><div id="yui_3_16_0_ym19_1_1542243121702_12657" dir="ltr"><span id="yui_3_16_0_ym19_1_1542243121702_12670">2. sip show registry shows it registered.</span></div><div id="yui_3_16_0_ym19_1_1542243121702_12658" dir="ltr"><span><br></span></div><div id="yui_3_16_0_ym19_1_1542243121702_12673" dir="ltr"><span id="yui_3_16_0_ym19_1_1542243121702_12672">Trunk allows for 2 channels which is not a problem here either</span><div id="yui_3_16_0_ym19_1_1542243121702_12671" dir="ltr"><br></div><span id="yui_3_16_0_ym19_1_1542243121702_12659">It's just weird that out of 4 queue member only 2 being called and log doesn't show anything else.</span><div></div><div id="yui_3_16_0_ym19_1_1542243121702_12661" dir="ltr"><span><br></span></div><div class="qtdSeparateBR" id="yui_3_16_0_ym19_1_1542243121702_12639"><br><br></div><div class="yahoo_quoted" id="yui_3_16_0_ym19_1_1542243121702_12627" style="display: block;"> <div id="yui_3_16_0_ym19_1_1542243121702_12626" style="font-family: Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div id="yui_3_16_0_ym19_1_1542243121702_12625" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;"> <div id="yui_3_16_0_ym19_1_1542243121702_12638" dir="ltr"> <font id="yui_3_16_0_ym19_1_1542243121702_12637" face="Arial" size="2"> <hr size="1" id="yui_3_16_0_ym19_1_1542243121702_12640"> <b id="yui_3_16_0_ym19_1_1542243121702_12663"><span id="yui_3_16_0_ym19_1_1542243121702_12662" style="font-weight: bold;">From:</span></b> "asterisk-users-request@lists.digium.com" <asterisk-users-request@lists.digium.com><br> <b><span style="font-weight: bold;">To:</span></b> asterisk-users@lists.digium.com <br> <b><span style="font-weight: bold;">Sent:</span></b> Thursday, November 15, 2018 11:20 AM<br> <b><span style="font-weight: bold;">Subject:</span></b> asterisk-users Digest, Vol 171, Issue 9<br> </font> </div> <div class="y_msg_container" id="yui_3_16_0_ym19_1_1542243121702_12624"><br><div dir="ltr">Send asterisk-users mailing list submissions to<br></div><div id="yui_3_16_0_ym19_1_1542243121702_12636" dir="ltr"> <a id="yui_3_16_0_ym19_1_1542243121702_12635" href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div><div id="yui_3_16_0_ym19_1_1542243121702_12634" dir="ltr"><br></div><div dir="ltr">To subscribe or unsubscribe via the World Wide Web, visit<br></div><div id="yui_3_16_0_ym19_1_1542243121702_12633" dir="ltr"> <a id="yui_3_16_0_ym19_1_1542243121702_12632" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div><div id="yui_3_16_0_ym19_1_1542243121702_12628" dir="ltr">or, via email, send a message with subject or body 'help' to<br></div><div dir="ltr"> <a href="mailto:asterisk-users-request@lists.digium.com" ymailto="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a><br></div><div id="yui_3_16_0_ym19_1_1542243121702_12623" dir="ltr"><br></div><div id="yui_3_16_0_ym19_1_1542243121702_12674" dir="ltr">You can reach the person managing the list at<br></div><div dir="ltr"> <a href="mailto:asterisk-users-owner@lists.digium.com" ymailto="mailto:asterisk-users-owner@lists.digium.com">asterisk-users-owner@lists.digium.com</a><br></div><div dir="ltr"><br></div><div dir="ltr">When replying, please edit your Subject line so it is more specific<br></div><div id="yui_3_16_0_ym19_1_1542243121702_12675" dir="ltr">than "Re: Contents of asterisk-users digest..."<br></div><div dir="ltr"><br></div><div dir="ltr"><br></div><div dir="ltr">Today's Topics:<br></div><div dir="ltr"><br></div><div dir="ltr"> 1. Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"> (Ivan Demkovitch)<br></div><div dir="ltr"> 2. Re: Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"> (Sebastian Nielsen)<br></div><div dir="ltr"> 3. Re: Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"> (Ivan Demkovitch)<br></div><div dir="ltr"> 4. Re: Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"> (Sebastian Nielsen)<br></div><div dir="ltr"><br></div><div dir="ltr"><br></div><div dir="ltr">----------------------------------------------------------------------<br></div><div dir="ltr"><br></div><div dir="ltr">Message: 1<br></div><div dir="ltr">Date: Thu, 15 Nov 2018 16:53:38 +0000 (UTC)<br></div><div dir="ltr">From: Ivan Demkovitch <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>><br></div><div dir="ltr">To: "<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>"<br></div><div dir="ltr"> <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br></div><div dir="ltr">Subject: [asterisk-users] Queue not dialing out to cell phone for some<br></div><div dir="ltr"> reason<br></div><div dir="ltr">Message-ID: <<a href="mailto:897612684.1161831.1542300818435@mail.yahoo.com" ymailto="mailto:897612684.1161831.1542300818435@mail.yahoo.com">897612684.1161831.1542300818435@mail.yahoo.com</a>><br></div><div dir="ltr">Content-Type: text/plain; charset="utf-8"<br></div><div dir="ltr"><br></div><div dir="ltr">Hello,<br></div><div dir="ltr">I have queues.conf setup with a group like so:<br></div><div dir="ltr">[Sales](StandardQueue)<br></div><div dir="ltr">announce = first<br></div><div dir="ltr">member => SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr">member => SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr">member => SIP/<a href="mailto:13145555555@callcentric" ymailto="mailto:13145555555@callcentric">13145555555@callcentric</a> ;Eric's cell<br></div><div dir="ltr">member => SIP/FF1565AABB2D-SLS ;Eric's Yealink<br></div><div dir="ltr">So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and this is what I see. Only 2 phones (internal) called. External <a href="mailto:SIP@callcentric" ymailto="mailto:SIP@callcentric">SIP@callcentric</a> is not being called.<br></div><div dir="ltr">Any idea why it's not being called?<br></div><div dir="ltr"><br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack<br></div><div dir="ltr"> Caller "aa" <15555555555> has entered the sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack<br></div><div dir="ltr"> -- Goto (queues,7001,1)<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack<br></div><div dir="ltr"> == "aa" <15555555555> entering sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000437 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000436 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000439 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000438 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-0000043b is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-0000043a is ringing<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'<br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">An HTML attachment was scrubbed...<br></div><div dir="ltr">URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/aec5fb8f/attachment-0001.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/aec5fb8f/attachment-0001.html</a>><br></div><div dir="ltr"><br></div><div dir="ltr">------------------------------<br></div><div dir="ltr"><br></div><div dir="ltr">Message: 2<br></div><div dir="ltr">Date: Thu, 15 Nov 2018 17:58:20 +0100<br></div><div dir="ltr">From: "Sebastian Nielsen" <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>><br></div><div dir="ltr">To: "'Ivan Demkovitch'" <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>>, "'Asterisk Users<br></div><div dir="ltr"> Mailing List - Non-Commercial Discussion'"<br></div><div dir="ltr"> <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br></div><div dir="ltr">Subject: Re: [asterisk-users] Queue not dialing out to cell phone for<br></div><div dir="ltr"> some reason<br></div><div dir="ltr">Message-ID: <000501d47d04$698e9480$3cabbd80$@sebbe.eu><br></div><div dir="ltr">Content-Type: text/plain; charset="utf-8"<br></div><div dir="ltr"><br></div><div dir="ltr">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.<br></div><div dir="ltr"><br></div><div dir="ltr">You need to go into android settings and make sure the SIP client is whitelisted in battery management.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Från: asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com" ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> För Ivan Demkovitch<br></div><div dir="ltr">Skickat: den 15 november 2018 17:55<br></div><div dir="ltr">Till: <a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div><div dir="ltr">Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Hello,<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">I have queues.conf setup with a group like so:<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">[Sales](StandardQueue)<br></div><div dir="ltr">announce = first<br></div><div dir="ltr">member => SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr">member => SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr">member => SIP/<a href="mailto:13145555555@callcentric" ymailto="mailto:13145555555@callcentric">13145555555@callcentric</a> ;Eric's cell<br></div><div dir="ltr">member => SIP/FF1565AABB2D-SLS ;Eric's Yealink<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.<br></div><div dir="ltr"><br></div><div dir="ltr">I did trace a call and this is what I see. Only 2 phones (internal) called. External <a href="mailto:SIP@callcentric" ymailto="mailto:SIP@callcentric">SIP@callcentric</a> is not being called.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Any idea why it's not being called?<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr"><br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack<br></div><div dir="ltr"> Caller "aa" <15555555555> has entered the sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack<br></div><div dir="ltr"> -- Goto (queues,7001,1)<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack<br></div><div dir="ltr"> == "aa" <15555555555> entering sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000437 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000436 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000439 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000438 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-0000043b is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-0000043a is ringing<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'<br></div><div dir="ltr"><br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">An HTML attachment was scrubbed...<br></div><div dir="ltr">URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment-0001.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment-0001.html</a>><br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">A non-text attachment was scrubbed...<br></div><div dir="ltr">Name: smime.p7s<br></div><div dir="ltr">Type: application/pkcs7-signature<br></div><div dir="ltr">Size: 5261 bytes<br></div><div dir="ltr">Desc: S/MIME Cryptographic Signature<br></div><div dir="ltr">URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment-0001.bin" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/f6682cf0/attachment-0001.bin</a>><br></div><div dir="ltr"><br></div><div dir="ltr">------------------------------<br></div><div dir="ltr"><br></div><div dir="ltr">Message: 3<br></div><div dir="ltr">Date: Thu, 15 Nov 2018 17:00:48 +0000 (UTC)<br></div><div dir="ltr">From: Ivan Demkovitch <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>><br></div><div dir="ltr">To: Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>>, 'Asterisk Users Mailing<br></div><div dir="ltr"> List - Non-Commercial Discussion' <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br></div><div dir="ltr">Subject: Re: [asterisk-users] Queue not dialing out to cell phone for<br></div><div dir="ltr"> some reason<br></div><div dir="ltr">Message-ID: <<a href="mailto:1273692324.1141360.1542301248670@mail.yahoo.com" ymailto="mailto:1273692324.1141360.1542301248670@mail.yahoo.com">1273692324.1141360.1542301248670@mail.yahoo.com</a>><br></div><div dir="ltr">Content-Type: text/plain; charset="utf-8"<br></div><div dir="ltr"><br></div><div dir="ltr">Sebastian,<br></div><div dir="ltr">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.Also, why I don't see anything in a log? I see only first 2 members being dialed. <br></div><div dir="ltr"><br></div><div dir="ltr"> From: Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>><br></div><div dir="ltr"> To: 'Ivan Demkovitch' <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>>; 'Asterisk Users Mailing List - Non-Commercial Discussion' <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> <br></div><div dir="ltr"> Sent: Thursday, November 15, 2018 10:58 AM<br></div><div dir="ltr"> Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"> <br></div><div dir="ltr">#yiv7898733751 #yiv7898733751 -- _filtered #yiv7898733751 {font-family:Helvetica;panose-1:2 11 6 4 2 2 2 2 2 4;} _filtered #yiv7898733751 {panose-1:2 4 5 3 5 4 6 3 2 4;} _filtered #yiv7898733751 {font-family:Calibri;panose-1:2 15 5 2 2 2 4 3 2 4;}#yiv7898733751 #yiv7898733751 p.yiv7898733751MsoNormal, #yiv7898733751 li.yiv7898733751MsoNormal, #yiv7898733751 div.yiv7898733751MsoNormal {margin:0cm;margin-bottom:.0001pt;font-size:11.0pt;font-family:sans-serif;}#yiv7898733751 a:link, #yiv7898733751 span.yiv7898733751MsoHyperlink {color:#0563C1;text-decoration:underline;}#yiv7898733751 a:visited, #yiv7898733751 span.yiv7898733751MsoHyperlinkFollowed {color:#954F72;text-decoration:underline;}#yiv7898733751 p.yiv7898733751msonormal0, #yiv7898733751 li.yiv7898733751msonormal0, #yiv7898733751 div.yiv7898733751msonormal0 {margin-right:0cm;margin-left:0cm;font-size:11.0pt;font-family:sans-serif;}#yiv7898733751 span.yiv7898733751E-postmall18 {font-family:sans-serif;}#yiv7898733751 .yiv7898733751MsoChpDefault {font-size:10.0pt;} _filtered #yiv7898733751 {margin:70.85pt 70.85pt 70.85pt 70.85pt;}#yiv7898733751 div.yiv7898733751WordSection1 {}#yiv7898733751 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.You need to go into android settings and make sure the SIP client is whitelisted in battery management. Från: asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com" ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> För Ivan Demkovitch<br></div><div dir="ltr">Skickat: den 15 november 2018 17:55<br></div><div dir="ltr">Till: <a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br></div><div dir="ltr">Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue)<br></div><div dir="ltr">announce = first<br></div><div dir="ltr">member => SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr">member => SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr">member => SIP/<a href="mailto:13145555555@callcentric" ymailto="mailto:13145555555@callcentric">13145555555@callcentric</a> ;Eric's cell<br></div><div dir="ltr">member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and this is what I see. Only 2 phones (internal) called. External <a href="mailto:SIP@callcentric" ymailto="mailto:SIP@callcentric">SIP@callcentric</a> is not being called. Any idea why it's not being called? <br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack<br></div><div dir="ltr"> Caller "aa" <15555555555> has entered the sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack<br></div><div dir="ltr"> -- Goto (queues,7001,1)<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack<br></div><div dir="ltr"> == "aa" <15555555555> entering sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000437 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000436 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000439 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000438 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-0000043b is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-0000043a is ringing<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">An HTML attachment was scrubbed...<br></div><div dir="ltr">URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/a0b9ed53/attachment-0001.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/a0b9ed53/attachment-0001.html</a>><br></div><div dir="ltr"><br></div><div dir="ltr">------------------------------<br></div><div dir="ltr"><br></div><div dir="ltr">Message: 4<br></div><div dir="ltr">Date: Thu, 15 Nov 2018 18:20:06 +0100<br></div><div dir="ltr">From: "Sebastian Nielsen" <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>><br></div><div dir="ltr">To: "'Ivan Demkovitch'" <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>>, "'Asterisk Users<br></div><div dir="ltr"> Mailing List - Non-Commercial Discussion'"<br></div><div dir="ltr"> <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br></div><div dir="ltr">Subject: Re: [asterisk-users] Queue not dialing out to cell phone for<br></div><div dir="ltr"> some reason<br></div><div dir="ltr">Message-ID: <001301d47d07$73aabf40$5b003dc0$@sebbe.eu><br></div><div dir="ltr">Content-Type: text/plain; charset="utf-8"<br></div><div dir="ltr"><br></div><div dir="ltr">Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred via data connection to the asterisk server.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Seems theres a problem with the trunk then.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">What does ”sip show registry” tell you?<br></div><div dir="ltr"><br></div><div dir="ltr">(asterisk -r in console and then sip show registry)<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">It should show a status of ”Registred” to your trunk operator.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Från: Ivan Demkovitch <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>> <br></div><div dir="ltr">Skickat: den 15 november 2018 18:01<br></div><div dir="ltr">Till: Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>>; 'Asterisk Users Mailing List - Non-Commercial Discussion' <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br></div><div dir="ltr">Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Sebastian,<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">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.<br></div><div dir="ltr"><br></div><div dir="ltr">Also, why I don't see anything in a log? I see only first 2 members being dialed. <br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr"> _____ <br></div><div dir="ltr"><br></div><div dir="ltr">From: Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a> <mailto:<a href="mailto:sebastian@sebbe.eu" ymailto="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>> ><br></div><div dir="ltr">To: 'Ivan Demkovitch' <<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a> <mailto:<a href="mailto:idemkovitch@yahoo.com" ymailto="mailto:idemkovitch@yahoo.com">idemkovitch@yahoo.com</a>> >; 'Asterisk Users Mailing List - Non-Commercial Discussion' <<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a> <mailto:<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> > <br></div><div dir="ltr">Sent: Thursday, November 15, 2018 10:58 AM<br></div><div dir="ltr">Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">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.<br></div><div dir="ltr"><br></div><div dir="ltr">You need to go into android settings and make sure the SIP client is whitelisted in battery management.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Från: asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com" ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" ymailto="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>> > För Ivan Demkovitch<br></div><div dir="ltr">Skickat: den 15 november 2018 17:55<br></div><div dir="ltr">Till: <a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a> <mailto:<a href="mailto:asterisk-users@lists.digium.com" ymailto="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> <br></div><div dir="ltr">Ämne: [asterisk-users] Queue not dialing out to cell phone for some reason<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Hello,<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">I have queues.conf setup with a group like so:<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">[Sales](StandardQueue)<br></div><div dir="ltr">announce = first<br></div><div dir="ltr">member => SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr">member => SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr">member => SIP/<a href="mailto:13145555555@callcentric" ymailto="mailto:13145555555@callcentric">13145555555@callcentric</a> ;Eric's cell<br></div><div dir="ltr">member => SIP/FF1565AABB2D-SLS ;Eric's Yealink<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.<br></div><div dir="ltr"><br></div><div dir="ltr">I did trace a call and this is what I see. Only 2 phones (internal) called. External <a href="mailto:SIP@callcentric" ymailto="mailto:SIP@callcentric">SIP@callcentric</a> is not being called.<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">Any idea why it's not being called?<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr"><br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack<br></div><div dir="ltr"> Caller "aa" <15555555555> has entered the sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:1@automated_attendant_normal" ymailto="mailto:1@automated_attendant_normal">1@automated_attendant_normal</a>:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack<br></div><div dir="ltr"> -- Goto (queues,7001,1)<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack<br></div><div dir="ltr"> == "aa" <15555555555> entering sales queue<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')<br></div><div dir="ltr"> -- Executing [<a href="mailto:7001@queues" ymailto="mailto:7001@queues">7001@queues</a>:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000437 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000436 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-00000439 is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-00000438 is ringing<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Nobody picked up in 30000 ms<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> -- Playing periodic announcement<br></div><div dir="ltr"> -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')<br></div><div dir="ltr"> -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF9EF375CCFC-SLS<br></div><div dir="ltr"> == Using SIP RTP CoS mark 5<br></div><div dir="ltr"> -- Called SIP/FF4C119EEBF8-SLS<br></div><div dir="ltr"> -- SIP/FF4C119EEBF8-SLS-0000043b is ringing<br></div><div dir="ltr"> -- SIP/FF9EF375CCFC-SLS-0000043a is ringing<br></div><div dir="ltr"> -- Stopped music on hold on SIP/callcentric15-00000435<br></div><div dir="ltr"> == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'<br></div><div dir="ltr"><br></div><div dir="ltr"> <br></div><div dir="ltr"><br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">An HTML attachment was scrubbed...<br></div><div dir="ltr">URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/28c3a2a7/attachment.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/28c3a2a7/attachment.html</a>><br></div><div dir="ltr">-------------- next part --------------<br></div><div dir="ltr">A non-text attachment was scrubbed...<br></div><div dir="ltr">Name: smime.p7s<br></div><div dir="ltr">Type: application/pkcs7-signature<br></div><div dir="ltr">Size: 5261 bytes<br></div><div dir="ltr">Desc: S/MIME Cryptographic Signature<br></div><div dir="ltr">URL: <<a 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