@rnrstar wrote:

We are running FreePBX 13.0.194.5
NAT = Yes, Static IP

We are experiencing lots of very high active calls anywhere from over a hundred to several hundred. This is displayed from the dashboard when looking at the statistics.

When looking at the full log file I see lots of the following:

[2018-04-16 08:41:26] DEBUG[35033][C-0000033e] format_wav.c: Skipping unknown block ‘LIST’
[2018-04-16 08:37:35] WARNING[34090][C-0000031e] chan_sip.c: This function can only be used on SIP channels.
[2018-04-16 08:37:35] ERROR[34030][C-000002ef] res_pjsip_header_funcs.c: This function requires a PJSIP channel.
[2018-04-16 08:37:35] WARNING[34092][C-0000031e] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
NOTICE[13090] chan_sip.c: Disconnecting call ‘SIP/TwilioIn4-00000705’ for lack of RTP activity in 31 seconds
[2018-04-16 08:37:49] WARNING[13090] chan_sip.c: Autodestruct on dialog ‘3dfce109341112224e7a60c8525d0e95@10.0.1.26:5060’ with owner SIP/1322-00000706 in place (Method: BYE). Rescheduling destruction for 10000 ms

Other symptoms we are seeing is when you place a call from the outside to the PBX, I hear the bleep and then a long period of silence and then eventually I hear the IVR announcement play.
When transferring a call, there is a long delay in the pickup.

This is a system that has been running very stable for quite some time. Two things have recently changed. We recently installed the datadog-agent v6 but we have since removed that agent. The other is that we moved the cdr and cel tables to another folder with more disk space. I had initially created a symlink to the whole directory but that apparently according to some docs I read was not a good thing to do. I have since put the .FMT files back to where they were and only symlinked the MYI and MYD files and initially that seemed to make everything run fine. After a couple days though, we are experiencing the issue again. I have checked the tables by running “check table cdr” and “check table cel” and they show as OK.

Any suggestions?

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@kendalldever15 wrote:

I have a PBXact with Sangoma S405 phones installed at a school and they added a phone to every class room. They had us add a button that a teacher can press that plays a “Code Red need Assistance” message which works just as they like, however the phone that activates the page also has the page come over it even though it is not in the page group.

What they are asking is the phone that activates the page stays silent so people in the room are unaware the activation has been done. Is this possible and if not is anyone out there want to give me a quote to develop this?

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@brk wrote:

Before I shoot myself in the foot, does anyone have input on the best way to manage call schedules with the calendar module?

I’ve read the documentation, I’ve linked my calendar, I’ve figured out a solution but I’m not sure if it is the best one.

What I currently have is a single calendar (ical format) with time blocks to specify the doctor on call - simply with a name in the calendar time slot.

I can control time conditions now based on having an appointment in a time slot. I can also make groups and select specific appointments within that calendar. But it looks like I have to login to freepbx and reselect which the appointments when they change.

I could also make 3 calendars, one for each person. And then put an event when they are on call.

Is there a better way to go about this? If someone has a link I’d happily read it. I wasn’t successful in finding a solution with web searching.

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@Hawkeye wrote:

Hi, having an issue with pjsip extensions that loose registration. The extensions have max 6 contacts. This problem is intermittent. The only fix is # fwconsole restart and the phones regain registration immediately.

Thought it might be firewall (iptables) but today disabled firewall and after a period of time ~ 40 minutes or so, phones loose registration. So its not iptables doing this.

Below from cli where bria extensions send registration packet but there is no response back to the phone.

<— Received SIP request (664 bytes) from TLS:xxx.241.183.250:48507 —>
OPTIONS sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:49770;branch=z9hG4bK-524287-1—f597d523438b6914;rport
Max-Forwards: 0
Contact: sip:308@192.168.25.17:49770;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:host.domain.tld:5061
From: "USER"sip:308@host.domain.tld:5061;tag=700ace02
Call-ID: 143678_rel51YTg4MWZhMmJmYmY0ZDljZjkzOWQ3NTBkYzAyMjFjOWQ
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build 99409
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (697 bytes) from TLS:xxx.241.183.250:48507 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:49770;branch=z9hG4bK-524287-1—76385e6ccaa3490b;rport
Max-Forwards: 70
Contact: sip:308@192.168.25.17:49770;rinstance=191f91d733894a81;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "USER"sip:308@host.domain.tld:5061
From: "USER"sip:308@host.domain.tld:5061;tag=0ca0eb57
Call-ID: 143678_rel51MzZjOTgyN2IwNGI3ZDZmYmExZTlkNGVhMDJlOTEwNDI
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build 99409
Content-Length: 0

All pjsip extensions use port 5061 tls.

There are two Bria softphones and they loose registration hourly.

Asterisk Version: 13.19.1
FreePBX Version: 10.13.66-22

Thanks for any assistance.

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@Fraser wrote:

Hello,

When trying to update the Process Management (pm2) module from 13.0.4.2 to 13.0.5 via the gui on one of our PBXes I get the following message:

image.png1465x830 85.4 KB

If I copy/paste all the text in the back box, then the full message is:

=$HOME/.node/share/man:$MANPATH &&
/var/www/html/admin/modules/pm2/node/node_modules/pm2/bin/pm2 ping" failed. Exit Code: 1(General error) Output: ================ Error Output: ================ module.js:338 throw err; ^ Error: Cannot find module ‘./internal/slice’ at Function.Module._resolveFilename (module.js:336:15) at Function.Module._load (module.js:278:25) at Module.require (module.js:365:17) at require (module.js:384:17) at Object. (/var/www/html/admin/modules/pm2/node/node_modules/pm2/node_modules/async/apply.js:15:14) at Module._compile (module.js:460:26) at Object.Module._extensions…js (module.js:478:10) at Module.load (module.js:355:32) at Function.Module._load (module.js:310:12) at Module.require (module.js:365:17)

/var/www/html/admin/libraries/Composer/vendor/symfony/process/Symfony/Component/Process/Process.php
public function mustRun($callback = null)
{
if ($this->isSigchildEnabled() && !$this->enhanceSigchildCompatibility) {
throw new RuntimeException(‘This PHP has been compiled with --enable-sigchild. You must use setEnhanceSigchildCompatibility() to use this method.’);
}

   if (0 !== $this->run($callback)) {
       throw new ProcessFailedException($this);
  }

I have also tried updating via command line using fwconsole ma downloadinstall pm2:

No repos specified, using: [standard,commercial] from last GUI settings

Starting pm2 download…
Processing pm2
Verifying local module download…Verified
Extracting…Done
Module pm2 successfully downloaded
Installing/Updating Required Libraries. This may take a while…The following me ssages are ONLY FOR DEBUGGING. Ignore anything that says ‘WARN’ or is just a war ning
Found npm-cache v0.7.0
Running installation…
[npm-cache] [INFO] using /home/asterisk/.package_cache as cache directory
[npm-cache] [INFO] [composer] Dependency config file /var/www/html/admin/modules /pm2/node/composer.json does not exist. Skipping install
[npm-cache] [INFO] [npm] config file exists
[npm-cache] [INFO] [npm] cli exists
[npm-cache] [INFO] [npm] hash of /var/www/html/admin/modules/pm2/node/package.js on: d533e5835beb0c379c4c571fb8ceaecc
[npm-cache] [INFO] [npm] cache exists
[npm-cache] [INFO] [npm] clearing installed dependencies at /var/www/html/admin/ modules/pm2/node/node_modules
[npm-cache] [INFO] [npm] …cleared
[npm-cache] [INFO] [npm] retrieving dependencies from /home/asterisk/.package_ca che/npm/2.15.11/d533e5835beb0c379c4c571fb8ceaecc.tar.gz
[npm-cache] [INFO] [bower] Dependency config file /var/www/html/admin/modules/pm 2/node/bower.json does not exist. Skipping install
[npm-cache] [INFO] [npm] done extracting
[npm-cache] [INFO] successfully installed all dependencies

Finished updating libraries!

[Symfony\Component\Process\Exception\ProcessFailedException]
The command “runuser ‘asterisk’ -s ‘/bin/bash’ -c ‘cd /var/www/html/admin/m
odules/pm2/node && mkdir -p /home/asterisk/.pm2 && mkdir -p /var/www/html/a
dmin/modules/pm2/node/logs && export HOME=/home/asterisk && export PM2_HOME
=/home/asterisk/.pm2 && export ASTLOGDIR=/var/log/asterisk && export ASTVAR
LIBDIR=/var/lib/asterisk && export PATH=$HOME/.node/bin:$PATH && export NOD
E_PATH=$HOME/.node/lib/node_modules:$NODE_PATH && export MANPATH=$HOME/.nod
e/share/man:$MANPATH && /var/www/html/admin/modules/pm2/node/node_modules/p
m2/bin/pm2 ping’” failed.
Exit Code: 1(General error)
Output:
================
Error Output:
================
module.js:338
throw err;
^
Error: Cannot find module ‘./internal/slice’
at Function.Module._resolveFilename (module.js:336:15)
at Function.Module._load (module.js:278:25)
at Module.require (module.js:365:17)
at require (module.js:384:17)
at Object. (/var/www/html/admin/modules/pm2/node/node_module
s/pm2/node_modules/async/apply.js:15:14)
at Module._compile (module.js:460:26)
at Object.Module._extensions…js (module.js:478:10)
at Module.load (module.js:355:32)
at Function.Module._load (module.js:310:12)
at Module.require (module.js:365:17)

ma [-f|–force] [-d|–debug] [–edge] [–color] [–skipchown] [-e|–autoenable] [–skipdisabled] [–snapshot="…"] [–format="…"] [-R|–repo="…"] [-t|–tag ="…"] [args1] … [argsN]

Results of fwconsole ma checkdepends pm2:

All dependencies met for module pm2

Resultes of fwconsole ma showupgrades:

No repos specified, using: [standard,commercial] from last GUI settings

Upgradable:
±-------±--------------±---------------+
| Module | Local Version | Online Version |
±-------±--------------±---------------+
| pm2 | 13.0.5 | 13.0.5 |
| xmpp | 13.0.17.9 | 13.0.17.13 |
±-------±--------------±---------------+

Results of fwconsole ma refreshsignatures:

Getting Data from Online Server…
Done
Checking Signatures of Modules…
Checking accountcodepreserve…
Good
Checking announcement…
Good
Checking arimanager…
Good
Checking asterisk-cli…
Good
Checking asteriskinfo…
Good
Checking backup…
Good
Checking blacklist…
Good
Checking builtin…
Signature Invalid
Could not find signed module on remote server!
Checking bulkdids…
Good
Checking bulkextensions…
Good
Checking bulkhandler…
Good
Checking callback…
Good
Checking callforward…
Good
Checking callrecording…
Good
Checking callwaiting…
Good
Checking campon…
Good
Checking cdr…
Good
Checking cel…
Good
Checking certman…
Good
Checking conferences…
Good
Checking configedit…
Good
Checking contactmanager…
Good
Checking core…
Good
Checking customappsreg…
Good
Checking cxpanel…
Good
Checking dashboard…
Good
Checking daynight…
Good
Checking dictate…
Good
Checking digium_phones…
Good
Checking digiumaddoninstaller…
Good
Checking disa…
Good
Checking donotdisturb…
Good
Checking endpoint…
Good
Checking extensionroutes…
Good
Checking fax…
Good
Checking featurecodeadmin…
Good
Checking findmefollow…
Good
Checking firewall…
Good
Checking framework…
Good
Checking fw_langpacks…
Good
Checking hotelwakeup…
Good
Checking iaxsettings…
Good
Checking infoservices…
Good
Checking irc…
Good
Checking ivr…
Good
Checking languages…
Good
Checking logfiles…
Good
Checking manager…
Good
Checking miscapps…
Good
Checking miscdests…
Good
Checking music…
Good
Checking outroutemsg…
Good
Checking paging…
Good
Checking parking…
Good
Checking phonebook…
Good
Checking phpinfo…
Good
Checking pinsets…
Good
Checking presencestate…
Good
Checking printextensions…
Good
Checking queueprio…
Good
Checking queues…
Good
Checking recordings…
Good
Checking restapi…
Good
Checking ringgroups…
Good
Checking setcid…
Good
Checking sipsettings…
Good
Checking soundlang…
Good
Checking superfecta…
Good
Checking sysadmin…
Good
Checking timeconditions…
Good
Checking tts…
Good
Checking ttsengines…
Good
Checking ucp…
Good
Checking ucpnode…
Good
Checking userman…
Good
Checking vmblast…
Good
Checking voicemail…
Good
Checking weakpasswords…
Good
Checking webrtc…
Good
Checking xmpp…
Good
Done
Updating Hooks…Done

This is a FreePBX distro install running version 13:

PBX Firmware: 10.13.66-16
PBX Service Pack: 1.0.0.0

Any help would be appreciated.

Many thanks,
Fraser

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@vespino wrote:

My FreePBX dropped it’s calls after 30 seconds, both inbound as outbound. I was able to fix the inbound calls by setting the correct local networks in SIP settings => NAT settings, but I’m struggling with the outbound calls. I have checked another FreePBX and applied the same settings, but without result.

screenshot.png746x278 15.1 KB

NAT is set to “Yes”.

Am I looking in the right place?

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@john2k wrote:

I am trying to get Camp-on to work on my FreePBX. I checked the advanced settings and checked the modules and it seems Camp-On is enabled. Below are screenshots, from the Advanced settings, Module Admin, and Feature Codes

When I dial a extension, the line is busy so it hangs up, straight after that I dial *82 it gives me a beep and hangs up. Then the user hangs up the phone it’s supposed to be that my phone rings so I can ring the user but it just doesn’t work.

Any ideas what I can do to fix this

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@PitzKey wrote:

Hello,

FreePBX Distro: 12.7.4-1804-1.sng7
Asterisk: 14.7.5
UCP: 14.0.2.5

We are using a certificate from GoDaddy
Ports: 443, 4443 & 4443 are open (for trusted addresses only.)

Phone says registered, but i cannot place or answer calls.
It rings when i have a incoming call, but it stops ringing when i click answer and nothing happens. Desk phone continues to ring, and caller hears the same.

See attached image - Console log:

image.png1758x172 14.7 KB

06:01:37 is when i tried to place a call.
06:02:34 is when it started ringing - incoming call
06:02:39 is when i pressed answer.

Anyone with a GoDaddy certificate has this working? or am i doing something wrong?

Thanks

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@XcamaroX wrote:

I would like to have a different ring when I transfer a call, so that the user knows it’s a transferred call and not a regular outside call.
I’m using the Yealink T23G phones and I know it offers custom ringtones.
Where in FreePBX can I find this feature to turn it on, or is there a code that I need to use?

Thanks in advance!

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@comtech wrote:

FreePBX/Asterisk 14

Is there a command that I can run from the CLI that will show how many voicemails are in a particular box, or if a particular voicemail has unread messages?

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@josep20 wrote:

Good Morning:

I recently applied a firewall update and since then on my dashboard i see the red flame with the following msg " Firewall service not running! " .

I went ahead and applied yum update to see if I was missing some upgrade (Iptables) but no luck. The problem persist.

Any help would be appreciated.

Thanks!

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@6LexLuthor6 wrote:

So here at the office I have two PBX’s one is FreePBX installed on Raspberry Pi with FreePBX v14.0.0.14 and Asterisk v13.20.0. The other is a Panasonic KX-NCP500. They are both on the same network. I believe I have a trunk between the two, but not sure if its configured properly. The reason I think its not configured properly is because when I call the Panasonic system from the FreePBX it goes strait to the voicemail at extension 101 no matter what extension I dial. If I call any extension of the FreePBX from the Panasonic I get the busy tone. No matter what I change in settings this happens or does not work at all. I will post how I have it configured below. I’m open to any and all suggestions. Thanks everyone!

This is how I have the FreePBX configured.

FreePbx Inbound Routes.JPG1580x670 65.6 KB

FreePbx Trunk sip settings outgoing.JPG1590x529 39.1 KB

FreePbx Trunk sip settings incoming.JPG1582x561 39.4 KB

FreePbx Trunk General.JPG1587x676 70.1 KB

FreePbx Outbound Routes Route Settings.JPG1587x746 94.2 KB

FreePbx Outbound Routes Dial Patterns.JPG1579x626 65.7 KB

FreePbx Trunk Dialed Number.JPG1607x769 114 KB

This is how I have the Panasonic PBX configured.

SIP Gateway Calling Party.JPG1453x586 127 KB

SIP Gateway Register.JPG1006x627 83.3 KB

SIP Gateway Main.JPG1246x684 112 KB

SIP Gateway Supplementary Service.JPG982x654 73.1 KB

This is a call log from FreePBX to the Panasonic system.

Call from Asterisk to Panasonic Part 1.JPG1190x967 255 KB

Call from Asterisk to Panasonic Part 2.JPG1110x980 278 KB

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@robertlo wrote:

I was wondering if anyone could help me with this question as I’m unable to find a good solution on the forums.

We’re looking into a FreePBX or PBXact appliance to replace our existing system. Retaining call recordings is a must have for us, and having access to the recordings in the GUI would make life easier for anyone who needs to pull this data.

I’m trying to find the best way to keep older recordings on the system. It looks like the archive files with the call recording reports module are a one way street, and trying to look through all the file names to find the specific call requested sounds like a nightmare, the it looks like the system prefers that you download the archive files manually… yikes!

I’ve tested mounting a NFS share on /var/spool/asterisk/monitor/ with pretty good results, but if the remote server is having downtime the recording halts the call until the server is back up. And it looks like FreePBX does not honnor the cache_record_files and record_cache_dir settings if you add them to asterisk.conf.

I could just rsync the files hourly and let FreePBX clean up the recordings, but I loose the nice GUI if I do that.

Has anyone found a good solution to this problem?

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@JimFromRamJack wrote:

I just got a call from a gov. agent in the Permits section of their division. Standard, day-to-day stuff that. This time, something weird happened, and I’d like to ask anyone reading this for help understanding what happened.

The call rang in on my desk phone as a co-worker “Bob Dobbins”, phone number = “30”. (Three-Zero, nothing more) I respond instantly when my co-workers call, so I didn’t waste a breath picking it up to find a kind, confused clerk from the Permits office of a local county regarding a job we’ve taken on. I’ve done the phone thing long enough to handle the call, but the weird number piqued my curiosity.

So I went to the Asterisk logs & pulled the entire SIP session up to the point I handed the call off to a Representative, to suss out wtf that was all about. Maybe it’s SOP for FOP or maybe it is a bug, I’m asking you.

The module pbx.c did its usual “executing …” steps & as usual executed “/var/www/html/admin/modules/superfecta/agi/superfecta.agi”. When the res_agi.c module launched the superfecta.agi script, this is what it logged:
CID Superfecta is Answering the Channel
CID Superfecta: Scheme is ALL
CID Superfecta: The DID passed from Asterisk is: 9045551212 (our DID for one remote office)
CID Superfecta: The number passed from Asterisk is: 30 (unedited except to trim timestamp)
CID Superfecta: The CID name passed from Asterisk is: 30 (unedited except to trim timestamp)
CID Superfecta: Executing Scheme…
CID Superfecta: CID Determined to be: ‘Bob Dobbins’ (of course that’s his real name – NOT)
CID Superfecta: Attempting to set lookupcid
CID Superfecta: Success!

Since Bob is a co-worker, the Dynamic Routes module sent the call to a Ring Group which is why it rang my desk. (In the server closet, by myself, I couldn’t hear any other extensions ring & being busy I didn’t take the time to check the ever-open FOP2 page.)

So I guess it’s safe to assume the gov’t agency pushed out the ‘30’ as their Caller ID, maybe; but what happened in FreePBX??

So seeing it was Superfecta, I went to Admin->CID Superfecta & found it just as I’d left it: with Data Source Name = FOP2 Phonebook then Asterisk Phonebook then FreePBX User Mgr & none others selected.

Scrolled down to “Test a phone number against the selected sources.” With the Debug Level set to ALL, I entered my own # to test, which got hits on all 3 sources as it should.

Now for the weird part:

I entered “30” (just three-zero) as the test phone number.

Bob Dobbins popped up in the results list from FOP2 Phonebook!!! He is in it, but has his proper cell number as Phone 1, nothing as Phone2. No “30”. I tried a couple of other non-phone-numbers & each one pulls up a name from our FOP2 Phonebook!!

I’m curious to know what logic superfecta.agi uses to pull names.

Anyone else get the same result? Admin->CID Superfecta with Data Source Name = FOP2 Phonebook then “Test a phone number against the selected sources.” with a non-phone number?

Anyone care to drop me a clue as to where I can make it stop doing that?? I’m hoping someone knows where to look in asterisk.visual_phonebook, or at least can say why this happens. It certainly lowers my expectations of superfecta.agi, and FreePBX in general.

Thank you for reading and thank you for any thoughtful answers.

J

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@precisionbs wrote:

We recently upgraded to FreePBX14, and now we are getting a “unsigned module” email notification. However, the dashboard doesn’t show an error. FreePBX “module admin” shows the signature as good. The two modules are “Digium Addons” and “Contact manager”

FreePBX 14.0.2.14
Current Asterisk Version: 13.19.1

Ran “fwconsole ma refreshsignatures” , and received the following error.

Checking cxpanel…
PHP Fatal error: Unsupported operand types in /var/www/html/admin/libraries/Console/Moduleadmin.class.php on line 1071
Whoops\Exception\ErrorException: Unsupported operand types in file /var/www/html/admin/libraries/Console/Moduleadmin.class.php on line 1071
Stack trace:

1. Whoops\Exception\ErrorException->() /var/www/html/admin/libraries/Console/Moduleadmin.class.php:1071

Any ideas?

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@bigjess007 wrote:

I am trying to figure out how to to answer or “seize” an inbound call, but I want it to keep ringing for the caller and have it keep ringing an extension until the caller hangs up.

Specifically, the call comes in, and is directed to a ring group. After 20 seconds I want the call to then be directed to a specific extension in the ring group and keep ringing that extension until the call hangs up.

The catch is at the 20 seconds I need the PBX to basically answer the call so the carrier doesn’t route it away but I don’t want the caller to realize the call has been answered so it needs to keep ringing for them, and then I need it to ring that specific extension until either it’s answered, or the caller hangs up.

I can’t seem to figure out how to do this. I’m sure there is a easy way but I’m struggling with it. Thanks!

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@bajramia wrote:

Hi All,
I just updated and it seems the call recording report stop working it would stay on loading please wait i had to downgrade from 14.0.1.7 to 14.0.1.3

Thank you

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@Hawkeye wrote:

Hi, intermittently all PJSIP endpoints loose registration for an unknown reason. The endpoints are sending REGISTER packets and there is no reply from asterisk. All the endpoints are sending to port 5061 for PJSIP.

The only immediate fix at this point is to run fwconsole restart and then PJSIP endpoints are able to register.

PBX Version: 10.13.66-18
Asterisk Version: 13.19.1

The output from cli below is from pjsip logger.

*CLI> pjsip set logger on
PJSIP Logging enabled
[2018-04-18 04:49:40] NOTICE[2743]: res_pjsip_transport_management.c:133 idle_sched_cb: Shutting down transport ‘TLS to xxx.241.183.250:33764’ since no request was received in 32 seconds
[2018-04-18 04:49:41] NOTICE[2743]: res_pjsip_transport_management.c:133 idle_sched_cb: Shutting down transport ‘TLS to xxx.241.183.250:33765’ since no request was received in 32 seconds
[2018-04-18 04:49:46] NOTICE[2743]: res_pjsip_transport_management.c:133 idle_sched_cb: Shutting down transport ‘TLS to xxx.241.183.250:33767’ since no request was received in 32 seconds
[2018-04-18 04:49:47] NOTICE[2743]: res_pjsip_transport_management.c:133 idle_sched_cb: Shutting down transport ‘TLS to xxx.241.183.250:33768’ since no request was received in 32 seconds
<— Received SIP request (576 bytes) from TLS:xxx.241.183.250:52331 —>
REGISTER sip:host.domain.tld:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.11:52331;branch=z9hG4bK17650f4b9BB69B4C
From: “User2” sip:260@host.domain.tld:5061;tag=F0B87703-10D0C384
To: sip:260@host.domain.tld:5061
CSeq: 1 REGISTER
Call-ID: eafdc0cb6a8127fab0957f3760884896
Contact: sip:260@192.168.25.11:52331;transport=tls;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomVVX-VVX_300-UA/5.7.0.11768
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<— Received SIP request (604 bytes) from TLS:xxx.241.183.250:33770 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33770;rport;branch=z9hG4bK4069725800
From: “User3” sip:270@host.domain.tld:5061;tag=1657078199
To: “User3” sip:270@host.domain.tld:5061
Call-ID: 3305533283@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:270@192.168.25.27:33770;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 5
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (655 bytes) from TLS:xxx.241.183.250:49539 —>
OPTIONS sip:xxx.64.49.61:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:40278;branch=z9hG4bK-524287-1—af91cb22da275f1f;rport
Max-Forwards: 0
Contact: sip:290@192.168.25.17:40278;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:xxx.64.49.61:5061
From: "User5"sip:290@xxx.64.49.61:5061;tag=85ac5b06
Call-ID: 143678_rel51YmY4NTIwMzllMzU2ZDM4NGUxOWUxMGJkNjEwMDg5ZTA
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (576 bytes) from TLS:xxx.241.183.250:52331 —>
REGISTER sip:host.domain.tld:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.11:52331;branch=z9hG4bK64d69bbdBE2EC19E
From: “User6” sip:290@host.domain.tld:5061;tag=29DD80C1-8482A862
To: sip:290@host.domain.tld:5061
CSeq: 1 REGISTER
Call-ID: fe8ec3c9283f255359db4ac31a884896
Contact: sip:290@192.168.25.11:52331;transport=tls;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomVVX-VVX_300-UA/5.7.0.11768
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<— Received SIP request (813 bytes) from TLS:xxx.241.183.250:2884 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.34:5062;branch=z9hG4bK1693511933;rport;alias
From: “Canyon DOOR BELL” sip:360@host.domain.tld:5061;tag=626827128
To: sip:360@host.domain.tld:5061
Call-ID: 1525333031-5062-1@BJC.BGI.CF.DE
CSeq: 2920 REGISTER
Contact: sip:360@192.168.25.34:5062;transport=tls
Authorization: Digest username=“360”, realm=“asterisk”, nonce=“1524024148/841d0e17bfb35edf2a22058396794fb3”, uri=“sip:host.domain.tld:5061”, response=“cbad31e6d058a2bd7b768ad55a20ea83”, algorithm=md5, cnonce=“14165908”, opaque=“1e668f9172758075”, qop=auth, nc=0000010d
Max-Forwards: 70
User-Agent: Grandstream HT-503 V2.0A 1.0.16.2 chip V2.2
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<— Received SIP request (604 bytes) from TLS:xxx.241.183.250:33772 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33772;rport;branch=z9hG4bK2122970758
From: “User3” sip:270@host.domain.tld:5061;tag=2580xxx.578
To: “User3” sip:270@host.domain.tld:5061
Call-ID: 4054958319@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:270@192.168.25.27:33772;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 5
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (573 bytes) from TLS:xxx.241.183.250:52331 —>
REGISTER sip:host.domain.tld:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.11:52331;branch=z9hG4bKca5a2def238671B0
From: “User1” sip:308@host.domain.tld:5061;tag=79D5EF17-E90AD458
To: sip:308@host.domain.tld:5061
CSeq: 1 REGISTER
Call-ID: 76f04b3eb9fa746469a8ab1cfa884896
Contact: sip:308@192.168.25.11:52331;transport=tls;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomVVX-VVX_300-UA/5.7.0.11768
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<— Received SIP request (664 bytes) from TLS:xxx.241.183.250:37036 —>
OPTIONS sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:48237;branch=z9hG4bK-524287-1—9dd49e43320bfa06;rport
Max-Forwards: 0
Contact: sip:308@192.168.25.17:48237;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:host.domain.tld:5061
From: "User1"sip:308@host.domain.tld:5061;tag=66fc2323
Call-ID: 143678_rel51NjU0NWQ2YjczODhkMjBjODA4MGQyNWY5NDg0MjQ3OTQ
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (697 bytes) from TLS:xxx.241.183.250:37036 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:48237;branch=z9hG4bK-524287-1—03577a3ca913a008;rport
Max-Forwards: 70
Contact: sip:308@192.168.25.17:48237;rinstance=d485742d5f05fb13;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "User1"sip:308@host.domain.tld:5061
From: "User1"sip:308@host.domain.tld:5061;tag=a3feab3c
Call-ID: 143678_rel51ZmZhYTVkNzVhZGI0MWY5MDk1NGM1MDc1YzVmMWU2MzU
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
Content-Length: 0

<— Received SIP request (661 bytes) from TLS:xxx.241.183.250:44092 —>
OPTIONS sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:46196;branch=z9hG4bK-524287-1—d786aa2fa0449331;rport
Max-Forwards: 0
Contact: sip:370@192.168.25.17:46196;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:host.domain.tld:5061
From: "BF"sip:370@host.domain.tld:5061;tag=93d6e747
Call-ID: 143678_rel51NzEzN2JiZTBjNmNhZTBkMzUwZGM1NTJlNzM0YjA2N2Q
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (691 bytes) from TLS:xxx.241.183.250:44092 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:46196;branch=z9hG4bK-524287-1—9e697d15f3d53d05;rport
Max-Forwards: 70
Contact: sip:370@192.168.25.17:46196;rinstance=42d1e608dfcc93ea;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "BF"sip:370@host.domain.tld:5061
From: "BF"sip:370@host.domain.tld:5061;tag=67ee3b4e
Call-ID: 143678_rel51ODU0MGU1ODQwMTg4MzViNTY0OTZjODIxYmRkNmMyNTc
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
Content-Length: 0

<— Received SIP request (664 bytes) from TLS:xxx.241.183.250:40643 —>
OPTIONS sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:41433;branch=z9hG4bK-524287-1—9b04b804cc80c068;rport
Max-Forwards: 0
Contact: sip:270@192.168.25.17:41433;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:host.domain.tld:5061
From: "User3"sip:270@host.domain.tld:5061;tag=b73f0614
Call-ID: 143678_rel51ZDRkYjIzNTM1ZGE3NTVhYTY4MzQ3NTA5Yzc4ZTFmNWY
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (697 bytes) from TLS:xxx.241.183.250:40643 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:41433;branch=z9hG4bK-524287-1—e3b0ff5f3a034938;rport
Max-Forwards: 70
Contact: sip:270@192.168.25.17:41433;rinstance=2bdf2ed9cfe1d851;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "User3"sip:270@host.domain.tld:5061
From: "User3"sip:270@host.domain.tld:5061;tag=ca154d5a
Call-ID: 143678_rel51ZDRhZDgxMjhkMGE4N2Y5ZWYwNWY0MjZhMDA0MmQ2OWU
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
Content-Length: 0

<— Received SIP request (691 bytes) from TLS:xxx.241.183.250:49539 —>
REGISTER sip:xxx.64.49.61:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:40278;branch=z9hG4bK-524287-1—1d691a05db4c3136;rport
Max-Forwards: 70
Contact: sip:290@192.168.25.17:40278;rinstance=9768295b066546b8;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "User5"sip:290@xxx.64.49.61:5061
From: "User5"sip:290@xxx.64.49.61:5061;tag=4b402f46
Call-ID: 143678_rel51NjQxZDVjMzlhNGU0MDFjOTY3NTNlMTFjNmVmZjk4NGM
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
Content-Length: 0

<— Received SIP request (667 bytes) from TLS:xxx.241.183.250:38748 —>
OPTIONS sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:38544;branch=z9hG4bK-524287-1—6dabf129e01e9673;rport
Max-Forwards: 0
Contact: sip:260@192.168.25.17:38544;ob;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”
To: sip:host.domain.tld:5061
From: "User2"sip:260@host.domain.tld:5061;tag=6ca4d31e
Call-ID: 143678_rel51NDhkZWI3NzhiMDc5ODllZTY4MTFmMjAyNWNmNDBhZjA
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
X-Connectivity-Probe-V4: 1
Content-Length: 0

<— Received SIP request (703 bytes) from TLS:xxx.241.183.250:38748 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.17:38544;branch=z9hG4bK-524287-1—54f6875bbd956836;rport
Max-Forwards: 70
Contact: sip:260@192.168.25.17:38544;rinstance=f04af7886dff2d02;transport=tls;+sip.instance=“urn:uuid:05b5636e-6711-59a5-8436-90474f4a762f”;reg-id=1
To: "User2"sip:260@host.domain.tld:5061
From: "User2"sip:260@host.domain.tld:5061;tag=16e6a77d
Call-ID: 143678_rel51MTNhNzI2YWQyMmQzODg4NGNiNGQ2NTNhYWNmYjcyNzY
CSeq: 1 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, SUBSCRIBE, MESSAGE
Supported: outbound, path
User-Agent: Bria Android 3.9.6 build xxx.09
Content-Length: 0

<— Received SIP request (606 bytes) from TLS:xxx.241.183.250:33773 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33773;rport;branch=z9hG4bK2078768585
From: “Noemie” sip:306@host.domain.tld:5061;tag=3074568277
To: “Noemie” sip:306@host.domain.tld:5061
Call-ID: 2466575000@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:306@192.168.25.27:33773;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 1
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (606 bytes) from TLS:xxx.241.183.250:33775 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33775;rport;branch=z9hG4bK2789563442
From: “Noemie” sip:306@host.domain.tld:5061;tag=2345946802
To: “Noemie” sip:306@host.domain.tld:5061
Call-ID: 2416007574@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:306@192.168.25.27:33775;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 1
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

[2018-04-18 04:50:00] NOTICE[2743]: res_pjsip_transport_management.c:133 idle_sched_cb: Shutting down transport ‘TLS to 24.114.69.134:25792’ since no request was received in 32 seconds
<— Received SIP request (641 bytes) from TLS:24.114.69.134:25793 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 25.122.35.34:56110;branch=z9hG4bK-524287-1—79839a68eeb68446;rport;alias
Max-Forwards: 70
Contact: sip:306@24.114.69.134:25824;rinstance=4a86e886c39c7xxx.;transport=tls
To: "Mama Noemie"sip:306@host.domain.tld:5061
From: "Mama Noemie"sip:306@host.domain.tld:5061;tag=48d5b91e
Call-ID: MzYxZTVjMjNkZWI5NGQ1MzQwYzI2NTZlYTcyZGQwOGU
CSeq: 542 REGISTER
Expires: 900
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Supported: outbound, path
User-Agent: Bria iOS release 3.9.7 stamp 38887.38893
Content-Length: 0

<— Received SIP request (610 bytes) from TLS:xxx.241.183.250:33776 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33776;rport;branch=z9hG4bK2700182291
From: “User5” sip:290@host.domain.tld:5061;tag=221426xxx.2
To: “User5” sip:290@host.domain.tld:5061
Call-ID: 1392017452@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:290@192.168.25.27:33776;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 2
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (610 bytes) from TLS:xxx.241.183.250:33778 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33778;rport;branch=z9hG4bK1825622623
From: “User5” sip:290@host.domain.tld:5061;tag=2972024860
To: “User5” sip:290@host.domain.tld:5061
Call-ID: 1900244065@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:290@192.168.25.27:33778;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 2
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (579 bytes) from TLS:xxx.241.183.250:52331 —>
REGISTER sip:host.domain.tld:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.11:52331;branch=z9hG4bK3a53d70587488C66
From: “USER4” sip:306@host.domain.tld:5061;tag=5D37E7A7-4956CBE8
To: sip:306@host.domain.tld:5061
CSeq: 1 REGISTER
Call-ID: 45ac294c6ece3e8582074e33aa884896
Contact: sip:306@192.168.25.11:52331;transport=tls;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomVVX-VVX_300-UA/5.7.0.11768
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<— Received SIP request (581 bytes) from TLS:xxx.241.183.250:52331 —>
REGISTER sip:host.domain.tld:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.25.11:52331;branch=z9hG4bK203550375FFF3378
From: “User3 Hosting” sip:270@host.domain.tld:5061;tag=DA3C7D5F-CAFE4220
To: sip:270@host.domain.tld:5061
CSeq: 1 REGISTER
Call-ID: 8fc28d518d8e1f472ebf4xxx.d3884896
Contact: sip:270@192.168.25.11:52331;transport=tls;methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”
User-Agent: PolycomVVX-VVX_300-UA/5.7.0.11768
Accept-Language: en
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<— Received SIP request (904 bytes) from TLS:xxx.241.183.250:33412 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.19:5060;branch=z9hG4bK2092412507;rport;alias
From: sips:290@host.domain.tld:5061;tag=20xxx.17282
To: sips:290@host.domain.tld:5061
Call-ID: 934525227-5060-1@BJC.BGI.CF.BJ
CSeq: 2157 REGISTER
Contact: sips:290@192.168.25.19:5060;transport=tls;reg-id=1;+sip.instance=“urn:uuid:00000000-0000-1000-8000-000B8295223A”
Authorization: Digest username=“290”, realm=“asterisk”, nonce=“1524024908/ef4b68e3b3b97394362cf8cc76bcdce6”, uri=“sip:host.domain.tld:5061”, response=“0b1fa16ae66d386b2500a6431dec16d1”, algorithm=md5, cnonce=“01645009”, opaque=“50ff02184d59f83d”, qop=auth, nc=00000069
X-Grandstream-PBX: true
Max-Forwards: 70
User-Agent: Grandstream GXP2130 1.0.9.69
Supported: path
Expires: 3600
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<— Received SIP request (609 bytes) from TLS:xxx.241.183.250:33779 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33779;rport;branch=z9hG4bK4086618053
From: “User2” sip:260@host.domain.tld:5061;tag=2228021749
To: “User2” sip:260@host.domain.tld:5061
Call-ID: 467109293@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:260@192.168.25.27:33779;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 3
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<— Received SIP request (608 bytes) from TLS:xxx.241.183.250:33781 —>
REGISTER sip:host.domain.tld:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.25.27:33781;rport;branch=z9hG4bK311735149
From: “User2” sip:260@host.domain.tld:5061;tag=3881730428
To: “User2” sip:260@host.domain.tld:5061
Call-ID: 551190160@192.168.25.27
CSeq: 1 REGISTER
Contact: sip:260@192.168.25.27:33781;transport=TLS
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink $DEVNAME 25.73.0.40
Expires: 3600
Line: 3
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

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@comtech wrote:

FreePBX 13.0.194.10
Asterisk 14.7.5
Queues 13.0.34.9
Queues Pro 13.0.25

All modules are up to date. We noticed a recent behavior where when a dynamic agent logs in via *45, the endpoint behaves correctly (i.e. the agent hears “Agent logged On/Off” and the call disconnects.), but the phone doesn’t ring with calls in queue. When we run a queue member count, we hear 0.

When we go to Reporting>Asterisk Info>Queues I see the agent signed into the queue (as expected), but I see the work “Invalid” on this string. As long as the agent shows as invalid, the queue will not send calls to the agent.

Q1 has 0 calls (max unlimited) in ‘leastrecent’ strategy (0s holdtime, 0s talktime), W:1, C:0, A:0, SL:0.0% within 60s
Members:
John Smith (Local/XXXXXXX@from-queue/n from hint:XXXXXXX@ext-local) (ringinuse enabled) (dynamic) (Invalid) has taken no calls yet

Logging out and in and rebooting the server doesn’t seem to help. Any ideas as to what is going on here?

Thanks for any insights.

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@coryallen wrote:

Hello,

Im getting the error:

Reload failed because retrieve_conf encountered an error: 255

On the main dashboard.

If I run fwconsole r --verbose via ssh I get:

Error(s) have occured, the following is the retrieve_conf output: exit: 255
Unable to continue. Call to undefined function FreePBX\modules\endpoint_apiApp() in /var/www/html/admin/modules/restapps/Restapps.class.php on line 1314
#0 /var/www/html/admin/libraries/Composer/vendor/filp/whoops/src/Whoops/Run.php(383): Whoops\Run->handleError(1, 'Call to undefin...', '/var/www/html/a...', 1314)
#1 [internal function]: Whoops\Run->handleShutdown()
#2 {main}

Any help to get past this error would be appreciated!

Thanks

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