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 <[email protected]>
To: 'Ivan Demkovitch' <[email protected]>; 'Asterisk Users Mailing List -
Non-Commercial Discussion' <[email protected]>
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
<[email protected]> För Ivan Demkovitch
Skickat: den 15 november 2018 17:55
Till: [email protected]
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@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@callcentric is not being called. Any idea why it's not being called?
-- Executing [1@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@automated_attendant_normal:2]
Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack
-- Goto (queues,7001,1)
-- Executing [7001@queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa"
<1555555> entering sales queue") in new stack
== "aa" <15555555555> entering sales queue
-- Executing [7001@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@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'
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
