Queue agent status always show "in use"

Asterisk Asterisk Support
1

Hi,
I have a problem with a queue agent (8200) whose status is always shown as “in use” (even if he is not taking a call).
Below is my queue configuration:

[general]
persistentmembers = yes
keepstats = yes
autofill = yes
autopause=no
maxlen = 0
monitor-format = wav
monitor-type = MixMonitor

[AtendedoresSCT]
strategy=rrmemory
servicelevel=45
timeout=30
joinempty=no
ringinuse=yes
autopause=no
member => SIP/8200
member => SIP/8201

The option ringinuse is enabled because of this problem, otherwise, the agent would not receive any calls (since it is constantly “in use”).

Call counter is also enabled in the SIP peers configuration:

[8200]
description=SCT 0
type=friend
secret=********
disallow=all
allow=alaw
host=dynamic
context=telefonia_ca
qualify=yes
callcounter=yes

And here is the dialplan configuration:

[telefonia_ca]
exten => 8200,1,Answer
exten => 8200,n,NoOp(Enviar chamadas para a fila)
exten => 8200,n,Queue(AtendedoresSCT,i,,,80)
exten => 8200,n,Hangup()

Can someone help me getting the agent status to show correctly so I can disable the ringinuse option?
I don’t have much experience with asterisk, but fire away any questions that might help solve the issue.
Asterisk version: Asterisk 11.7.0~dfsg-1ubuntu1
Thanks in advance.

2

If you do a core show channels does that extension have a channel stuck up that’s counting against it’s call counter?

3

At the moment my queue looks like this:

AtendedoresSCT has 0 calls (max unlimited) in 'rrmemory' strategy (12s holdtime, 54s talktime), W:0, C:7307, A:804, SL:95.3% within 45s
   Members: 
      SIP/8200 (ringinuse enabled) (In use) has taken 7307 calls (last was 1275 secs ago)
      SIP/8201 (ringinuse enabled) (Unavailable) has taken no calls yet
   No Callers

As you can see, despite having no callers, 8200 is “in use”.

Regarding the core show channels, this is the output I get:

UbuntuAsterisk*CLI> core show channels 
Channel              Location             State   Application(Data)             
SIP/8200-000028f8    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-00003b58  (None)               Up      AppDial((Outgoing Line))      
SIP/8200-00004537    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-000032b3  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000022eb    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-000022b 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00001bd7    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000280    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000016b8    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-00000a9 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000022b1    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000a9a    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-000021a 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/LLoios-00003dc6  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-0000420f    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003dbe    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003dc7    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000021a7    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-0000056 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00003de2    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003b4d    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00004467    113@telefonia_ca:2   Up      Dial(SIP/RSJoao,12,r)         
SIP/LLoios-000039c0  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-00000cc 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RSJoao-000044c0  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RFlores-000038ef 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00003b57    113@telefonia_ca:2   Up      Dial(SIP/RSJoao,12,r)         
SIP/R.S.Joao-0000158 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-00000f8 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-0000056b    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RFlores-00003b4c 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00000ccb    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-000000a 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RSJoao-00003837  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Caldeireiros-0 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00001583    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000f8d    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000000a4    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000044c1    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-00001b7 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-000031b 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Carmelitas-000 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00001b77    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Caldeireiros-0 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Flores-000004f 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000031bd    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-00003f97  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00000be5    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000038e4    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000909    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-000038e3  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000044 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-00002db 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-00001cb 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Flores-0000090 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000004fe    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-0000129 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Flores-0000138 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000038f0    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-00000bc 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/LLoios-000028f7  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00002dbf    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/Vit_Mouzinho-000 (None)               Up      AppDial((Outgoing Line))      
SIP/8200-00001cbf    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000032ec    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003838    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Caldeireiros-0 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RFlores-000032eb 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000048 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RSJoao-00003dbd  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000033 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-0000129b    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000450    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00000bc3    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00002d29    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-00002d5 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000173 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/LLoios-00002754  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RSJoao-00003de1  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-000008f 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00000335    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00002410    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00002de9    108@telefonia_ca:2   Up      Dial(SIP/Vit_Mouzinho,12,r)   
SIP/8200-0000138b    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00002d54    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000032b4    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-000017e 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00000490    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-0000173d    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000008fc    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003f98    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Ferraz-000022e 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-000017c 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.Flores-00000b2 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00002440    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-0000113 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-000012c 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/LLoios-0000243f  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000104 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/RSJoao-00004313  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000017c5    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RFlores-00004535 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00001157    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-0000113c    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-000012cc    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00001049    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-0000178 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00002755    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Flores-00001bd 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000003e6    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-00004468  (None)               Up      AppDial((Outgoing Line))      
SIP/R.S.Joao-0000077 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/R.S.Joao-0000027 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/LLoios-0000240f  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00004314    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/RSJoao-00003a46  8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-0000178b    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.S.Joao-0000307 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000017f0    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003a47    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Flores-000016b 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-000039c6    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-0000077d    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/R.Ferraz-0000420 8200@telefonia_ca:3  Up      Queue(AtendedoresSCT,i,,,60)  
SIP/8200-00000b28    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
SIP/8200-00003079    8200@telefonia_ca:1  Up      AppQueue((Outgoing Line))     
126 active channels
63 active calls
8663 calls processed

It shows that there are 63 active calls, but I am positive that there are none. Unless active calls doesn´t mean ongoing calls… (sorry for the stupid question).
Am I not ending my calls right?

4

I suggest check the device state, with DEVICE_STATE function to see if state change is working and try the busylevel option

The busylevel option only works if call counters are enabled via the above option. If call counters are enabled, then busylevel allows you to set a threshold for when to consider this device busy. If busylevel is set to 2, then only at 2 or more calls will the device state report BUSY. The busylevel option can only be set for peers.

5

I think you found your problem.

Your peer 8200 has a lot of those channels up, You can hang up each call one at a time or restart asterisk to clear them all at once.

6

Ok, but, after closing the channels, how do I make sure that this situation doesn´t happen again?

7

You could look at SIP Session timers, RTP Timeout or good old fashioned Set(TIMEOUT(absolute)

8

set the ringinuse option to no, make sure callcounter option is set yes and then set busylevel to 1

SIP timer inform the signaling layers (SIP) when RTP has stopped flowing. The Initial Media Inactivity timer indicates that no RTP was ever received on the IP channel. The Media Inactivity Timer indicates that RTP has stopped flowing. When the signaling layers receive indications that these timers have expired, the signaling releases the channel. This feature resolves the problem where signaling gets out of sync and IP channels remain up with no one on the other side of the call.

9

Thanks for the reply!
I set the ringinuse option to no and the busylevel to 1. Let´s see how it behaves from here on.

Despite that, I’m a bit confused by the last part of your answer (because I’m a noob). Are you saying that the SIP session timer of this peer should be set to a lower value?
Right now, these are the configurations of the 8200 peer:

* Name       : 8200
  Description  : SCT 0
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : telefonia_ca
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : 
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Busy level   : 1
  Dynamic      : Yes
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Force rport  : Auto (No)
  Symmetric RTP: No
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : (null)
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 
  SIP Options  : (none)
  Codecs       : (alaw)
  Codec Order  : (alaw:20)
  Auto-Framing :  No 
  Status       : UNKNOWN
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No

The SIP session is set for 1800 seconds. Should it be lower?

10

session-expires -is the maximum session refresh interval in seconds. Defaults to 1800 secs. to it is fine.

check the http://svn.digium.com/svn/asterisk/trunk/configs/samples/sip.conf.sample there is session called RTP timers & SIP Session-Timers , this will help you to get a better understanding

11

From what I understand from the link you posted, RTP timeout hangs up a call if there is no audio in the channel during a certain amount of time.

My problem might actually be related to that, because the phones that call 8200 are placed in various streets, so there is always going to be noise going in the channel (from cars and such). On top of that, sometimes the 8200 phone becomes unreachable to Asterisk and some seconds later reachable again.

So, could the problem be that a call from a street is taking place, the 8200 phone becomes unreachable, and so the channels for that call are never closed because there is always audio in them? If so, could the RTP-timeout be set to just one end of the call?

12

Without any log file or cli I’m not quite sure what your problem could be but you can try this

rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activityon the audio channel

13

I don´t see a lot of errors in the CLI, but one that pops up sometimes this:

  == Using SIP RTP CoS mark 5
    -- Executing [8200@telefonia_ca:1] Answer("SIP/RSJoao-00000043", "") in new stack
    -- Executing [8200@telefonia_ca:2] NoOp("SIP/RSJoao-00000043", "Enviar chamadas para a fila") in new stack
    -- Executing [8200@telefonia_ca:3] Queue("SIP/RSJoao-00000043", "AtendedoresSCT,i,,,80") in new stack
    -- Started music on hold, class 'default', on SIP/RSJoao-00000043
  == Using SIP RTP CoS mark 5
[Aug 31 23:06:58] WARNING[8292]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 57d18c576f86fa996c5bed086587da5e@192.168.60.241:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Aug 31 23:06:58] WARNING[8292]: chan_sip.c:4204 retrans_pkt: Hanging up call 57d18c576f86fa996c5bed086587da5e@192.168.60.241:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Nobody picked up in 7000 ms
  == Using SIP RTP CoS mark 5
[Aug 31 23:07:06] NOTICE[8292]: chan_sip.c:29427 sip_poke_noanswer: Peer '8200' is now UNREACHABLE!  Last qualify: 16
[Aug 31 23:07:09] WARNING[8292]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 0384deb91e709cba0e6487d4551152fe@192.168.60.241:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Aug 31 23:07:09] WARNING[8292]: chan_sip.c:4204 retrans_pkt: Hanging up call 0384deb91e709cba0e6487d4551152fe@192.168.60.241:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Nobody picked up in 6000 ms
    -- Stopped music on hold on SIP/RSJoao-00000043
    -- Started music on hold, class 'default', on SIP/RSJoao-00000043
    -- Stopped music on hold on SIP/RSJoao-00000043
    -- Executing [8200@telefonia_ca:4] Hangup("SIP/RSJoao-00000043", "") in new stack
  == Spawn extension (telefonia_ca, 8200, 4) exited non-zero on 'SIP/RSJoao-00000043'

Here I have a call entering the queue and right after that the first warning pops up (line 7), after that the call (I don´t know which call) is hanged up.
After that, on line 12, the 8200 peer becomes unreachable.

From what I understand, because 8200 is a peer, sometimes asterisk will poke it. As it didn´t reply in time it was considered unreachable.

What I don´t understand are the other errors, the retrans_pkt ones. I don´t know what calls are being hanged up. How can I check what call is 57d18c576f86fa996c5bed086587da5e.
The 192.168.60.241 address is the asterisk server’s address.

At the end of the day, and after checking the link in the warning, I am leaning towards this being a hardware fault with the 8200 phone.

14

there is a connectivity issue , if you read the log carefully you will see asterisk is giving you a reference link https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

15

Yup. I think it is the phone’s fault. It´s time to replace it :). Thanks for all your help!