[asterisk-users] Queue not dialing out to cell phone for some reason

[Ivan Demkovitch]

Sebastian,
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. 

      From: Sebastian Nielsen <sebastian at sebbe.eu>
 To: 'Ivan Demkovitch' <idemkovitch at yahoo.com>; 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users at lists.digium.com> 
 Sent: Thursday, November 15, 2018 10:58 AM
 Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some reason
   
#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 <asterisk-users-bounces at lists.digium.com> För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: asterisk-users at lists.digium.com
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)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/13145555555 at callcentric ;Eric's cell
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 SIP at callcentric is not being called.  Any idea why it's not being called?  
    -- Executing [1 at automated_attendant_normal:1] Verbose("SIP/callcentric15-00000435", "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack
  Caller "aa" <15555555555> has entered the sales queue
    -- Executing [1 at automated_attendant_normal:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
    -- Goto (queues,7001,1)
    -- Executing [7001 at queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack
  == "aa" <15555555555> entering sales queue
    -- Executing [7001 at queues:2] BackGround("SIP/callcentric15-00000435", "/etc/asterisk/automated-attendant-prompts/aa_sales") in new stack
    -- <SIP/callcentric15-00000435> Playing '/etc/asterisk/automated-attendant-prompts/aa_sales.slin' (language 'en')
    -- Executing [7001 at queues:3] Queue("SIP/callcentric15-00000435", "sales,,,,85") in new stack
    -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-00000437 is ringing
    -- SIP/FF9EF375CCFC-SLS-00000436 is ringing
    -- Nobody picked up in 30000 ms
    -- Nobody picked up in 30000 ms
    -- Stopped music on hold on SIP/callcentric15-00000435
    -- Playing periodic announcement
    -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')
    -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-00000439 is ringing
    -- SIP/FF9EF375CCFC-SLS-00000438 is ringing
    -- Nobody picked up in 30000 ms
    -- Nobody picked up in 30000 ms
    -- Stopped music on hold on SIP/callcentric15-00000435
    -- Playing periodic announcement
    -- <SIP/callcentric15-00000435> Playing 'queue-periodic-announce.ulaw' (language 'en')
    -- Started music on hold, class 'default', on channel 'SIP/callcentric15-00000435'
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF9EF375CCFC-SLS
  == Using SIP RTP CoS mark 5
    -- Called SIP/FF4C119EEBF8-SLS
    -- SIP/FF4C119EEBF8-SLS-0000043b is ringing
    -- SIP/FF9EF375CCFC-SLS-0000043a is ringing
    -- Stopped music on hold on SIP/callcentric15-00000435
  == Spawn extension (queues, 7001, 3) exited non-zero on 'SIP/callcentric15-00000435'

   
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181115/a0b9ed53/attachment.html>