11 thoughts on - Queue Not Dialing Out To Cell Phone For Some Reason

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  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 För Ivan Demkovitch Skickat: den 15 november 2018 17:55
  Till: asterisk-users@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@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
  — 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
  — 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
  — 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’

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• 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
  To: ‘Ivan Demkovitch’ ; ‘Asterisk Users Mailing List – Non-Commercial Discussion’
  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 För Ivan Demkovitch Skickat: den 15 november 2018 17:55
  Till: asterisk-users@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@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
      — 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
      — 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
      — 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’

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  Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred via data connection to the asterisk server.

  Seems theres a problem with the trunk then.

  What does ”sip show registry” tell you?

  (asterisk -r in console and then sip show registry)

  It should show a status of ”Registred” to your trunk operator.

  Från: Ivan Demkovitch
  Skickat: den 15 november 2018 18:01
  Till: Sebastian Nielsen ; ‘Asterisk Users Mailing List – Non-Commercial Discussion’
  Ämne: Re: SV: [asterisk-users] Queue not dialing out to cell phone for some reason

  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 >
  To: ‘Ivan Demkovitch’ >; ‘Asterisk Users Mailing List – Non-Commercial Discussion’ >
  Sent: Thursday, November 15, 2018 10:58 AM
  Subject: SV: [asterisk-users] Queue not dialing out to cell phone for some reason

  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 > För Ivan Demkovitch Skickat: den 15 november 2018 17:55
  Till: asterisk-users@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@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
  — 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
  — 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
  — 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’

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• what does the output of ‘queue show sales’ show?

  Do you have queue logging enabled? Have you looked in the queue log to see what events are firing?

• John, This is output of command below. How do I enable and log queue events?The 1555@callcentric is the one I’m curious about. I just tried calling into “sales” again and it didn’t change this “last was 1219067” output Sales has 0 calls (max unlimited) in ‘ringall’ strategy (9s holdtime, 156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
     Members:
        SIP/15555555555@callcentric (ringinuse disabled) (Not in use) has taken 4 calls (last was 1219067 secs ago)
        SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls yet
        SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
        SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls yet
     No Callers

  From: John Kiniston
  To: idemkovitch@yahoo.com; Asterisk Users Mailing List – Non-Commercial Discussion
  Sent: Thursday, November 15, 2018 2:21 PM
  Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason

  what does the output of ‘queue show sales’ show?

  Do you have queue logging enabled? Have you looked in the queue log to see what events are firing?

  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
      — 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
      — 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

• From: John Kiniston
  To: idemkovitch@yahoo.com Sent: Thursday, November 15, 2018 3:17 PM
  Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason

  OK.

  So it looks like asterisk can’t ring FF1565AABB2D-SLS because it’s invalid.

  is the user at  ‘15555555555’ actually able the answer calls? I wouldn’t expect that agent to work configured that way, I’d use a LOCAL channel to direct the call to a context that sets the call up before dialing out.

  You configure queue logging in logger.conf , Look at the settings queue_log = yes queue_log_to_file = yes queue_log_name = queue_log

  John, This is output of command below. How do I enable and log queue events?The 1555@callcentric is the one I’m curious about. I just tried calling into “sales” again and it didn’t change this “last was 1219067” output Sales has 0 calls (max unlimited) in ‘ringall’ strategy (9s holdtime, 156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
     Members:
        SIP/15555555555@callcentric (ringinuse disabled) (Not in use) has taken 4 calls (last was 1219067 secs ago)
        SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls yet
        SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
        SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls yet
     No Callers

  [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

• John, FF1565AABB2D-SLS is probably invalid because it’s not registered/lost registration. This client is connected via VPN to our network, it usually works when it’s “warm”. Not concerned about it too much.
  15555555555@callcentric OTOH is an actual cell phone that should be dialed out via callcentric trunk. Maybe I’m smoking something thinking it was working before. I know it works from extensions.conf ————————-[globals]
  ERIC_CELL=SIP/15555555555@callcentric… exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
          same => n,VoiceMail(105@default,u)
  ———————————

• My settings for the queue.log are in the [general] section of logger.conf

  I’m running 13, I didn’t see what version you said you were running.

  If I wanted to add a LOCAL channel to my queue I’d do it as

  member => LOCAL/7124@kiniston-intern,0,John,hint:7124@kiniston-intern

• John, Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and configured as per book.. So, from what I understand – LOCAL means I want local extension to be a member of a queue. For example, I have this:
  [internal]
  ;Eric on extension 105
  exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
          same => n,VoiceMail(105@default,u)
  ———————-

• Got it working! Thanks a lot again. As a bonus, is there a background on why SIP/ did not work with a sip trunk provider? 🙂

  From: John Kiniston
  To: Ivan Demkovitch
  Cc: Asterisk Users Mailing List – Non-Commercial Discussion
  Sent: Friday, November 16, 2018 3:08 PM
  Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some reason

  So, LOCAL in this context is a ‘Technology’ or ‘Channel Driver’ , Instead of PJSIP, SIP, IAX, it’s sending a call to a dialplan target.

  Your entry in queues.conf with LOCAL/105@internal would send the call to the context ‘internal’ extension ‘105’ and execute whatever that dialplan does.

  The parameters I gave are actually part of the Queue member definition,

  From the example queues.conf:

   Each member of this call queue is listed on a separate line in
  ; the form technology/dialstring.  “member” means a normal member of a
  ; queue.  An optional penalty may be specified after a comma, such that
  ; entries with higher penalties are considered last.  An optional member
  ; name may also be specified after a second comma, which is used in log
  ; messages as a “friendly name”.  Multiple interfaces may share a single
  ; member name. An optional state interface may be specified after a third
  ; comma. This interface will be the one for which app_queue receives device
  ; state notifications, even though the first interface specified is the one
  ; that is actually called.
  ;
  ; A hint can also be used in place of the state interface using the format
  ; hint:@. If no context is specified then ‘default’ will
  ; be used.

  So 0 is the Penalty for the user Then ‘eric’ is the Member name and the state interface is using the hint defined for the user.

  John, This is output of command below. How do I enable and log queue events?The 1555@callcentric is the one I’m curious about. I just tried calling into “sales” again and it didn’t change this “last was 1219067” output Sales has 0 calls (max unlimited) in ‘ringall’ strategy (9s holdtime, 156s talktime), W:0, C:4, A:6, SL:0.0% within 0s
     Members:
        SIP/15555555555@callcentric (ringinuse disabled) (Not in use) has taken 4 calls (last was 1219067 secs ago)
        SIP/FF4C119EEBF8-SLS (ringinuse disabled) (Not in use) has taken no calls yet
        SIP/FF1565AABB2D-SLS (ringinuse disabled) (Invalid) has taken no calls yet
        SIP/FF9EF375CCFC-SLS (ringinuse disabled) (Not in use) has taken no calls yet
     No Callers

  [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, LOCAL in this context is a ‘Technology’ or ‘Channel Driver’ , Instead of PJSIP, SIP, IAX, it’s sending a call to a dialplan target.

  Your entry in queues.conf with LOCAL/105@internal would send the call to the context ‘internal’ extension ‘105’ and execute whatever that dialplan does.

  The parameters I gave are actually part of the Queue member definition,

  From the example queues.conf:

  Each member of this call queue is listed on a separate line in
  ; the form technology/dialstring. “member” means a normal member of a
  ; queue. An optional penalty may be specified after a comma, such that
  ; entries with higher penalties are considered last. An optional member
  ; name may also be specified after a second comma, which is used in log
  ; messages as a “friendly name”. Multiple interfaces may share a single
  ; member name. An optional state interface may be specified after a third
  ; comma. This interface will be the one for which app_queue receives device
  ; state notifications, even though the first interface specified is the one
  ; that is actually called.
  ;
  ; A hint can also be used in place of the state interface using the format
  ; hint:@. If no context is specified then ‘default’ will
  ; be used.

  So 0 is the Penalty for the user Then ‘eric’ is the Member name and the state interface is using the hint defined for the user.