@tim007 wrote:

We just migrated from FreePBX 13 to FreePBX 14, and a user asked me how, in the new UCP, to view the time that a voicemail was left. I went looking and am not finding a way. Oddly, the Wiki shows an image of the UCP without the time show, but then says:

https://wiki.freepbx.org/display/FPG/UCP+Voicemail
"For each voicemail we can see the following;
Date- Date voicemail was left
Time- Time voicemail was left
CID- Caller ID of who left the voicemail if supplied
Mailbox- Extension number where voicemail was left
Length- Length in Mins and Seconds of the message
Controls- For each message we can choose any of the following Controls"

There is a place to change what columns are shown, but they are all already checked, and the list is shorter than what is noted in the wiki:

• Date
• CID
• Playback
• Duration
• Controls

Is there some setting somewhere that I am missing that enables the additional columns?

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@georgedal wrote:

I’ve been using FreePBX (PBX Firmware:10.13.66-17) for a couple of months and I can say that I’m really satisfied. Recently I made some new settings on how the calls will be routed in case that I don’t answer my sip extensions.

So if a call is coming from my DID to my extension (lets say 201) I will see the Caller ID correctly. If I don’t answer, the call will be forwarded to my cellphone but it will have an invalid number because the country code is stripped of by my DID provider. I asked them to alter my account setting on that but that was not an option.

What I need is a dialplan that will detect when an incoming number starts with 2X or 69X and add my country prefix (30) without the plus sign. I didn’t find something that worked in my case.

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@dsstoronto wrote:

Hello,

I have a problem, our office uses a analog PBX phone system, we have a small remote office where we need to extend the phone system, with the extensions and such, we already have a VPN tunnel, I am wondering if there is a way to install free PBX and get the sangoma phones for the remote office. We would install the server at the main office and some device that would plug in to the analog PBX. then we can put the phones on the remote site and have the extensions extended.

What would the best way be?

Any options are appreciated.

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@spikeingtons wrote:

Hi All,

I have a central Asterisk phone book that was created with speed dials (imported from CSV)

However, when dialing the speed dial (eg, 777) it tries to call externally!

Is there something silly I am missing?

image.png1452x99 1.61 KB

image.png1444x37 1.9 KB

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@iseb wrote:

Hello.

I’m trying to register a “trunk”, or mostly my provider SIP account with SIP credentials.
I can successfully setup the account on my phones or VOIP applications, but not in FreePBX.
My provider of course don’t make any support.

The problem, I think, is that the registration string I should use do not work as expected. This is because the username of the account contains the @ sign.

In the log, I have, a timeout registering to the wrong serveur, or a fail with the good server (and bad username).

I tried a lot of things without success. My provider don’t answer any question.

Have anyone a tip to register this correctly ? Or without the registry string ?
I’m kind of blocked. Thanks

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@TechPlus wrote:

I have a freepbx that is acting strange. Every few hours it will stop accepting calls or wont allow calls to go out. I am using a TDM-800p card with HWEC. I’m not sure if it was hacked or what. One strange thing that I noticed is the BLF for an extension on one of the aastra phones will be flashing like its constantly ringing when it isn’t ringing. it is a |PBX Firmware:|5.211.65-11||PBX Service Pack:|1.0.0.0|

doing an amportal restart will fix the issues temporarily.

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@rnrstar wrote:

Using FreePBX 13.0.192.19 I have a couple extensions linked to an Active Directory user account that keeps dropping the link. I have all the other users working fine but one or two users keeps losing the link to the active directory account. When I go the extension for either of those users, the field “Link to a Different Default User” gets set to “None.”

Any suggestions as to what to look for?

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@dijkstra wrote:

Hi,

We have a small problem, when calling a specific number on our PBX (from a external phone) all the mobile phones are ringing. And if we pickup press the one it is ok. but when we let the call go on and ring until it stops and the PBX calls us again for the second time or third time and we pickup we cannot get the call. it will state that there is no call. But the callee is still waiting for a pickup?

Hope you can help!

Aron

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@jessy5765 wrote:

We have 2 VLAN’s for the network:

VLAN 30 (Data): Untagged
VLAN 20 (Voice): Tagged

If I am using option 66 to program the phones do I need to set up the PBX to first accept connections on VLAN30 in order to provision the phones and then have an additional eth port on VLAN20 accepting the voice traffic?

Is there any other way or is this the “correct” way.

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@kwriley87 wrote:

Hi there

I am using FreePBX Distro 14.

We have a small call center and would like to pull call metrics, mainly how many inbound calls there was each day, how many calls each extension took, etc.

I know there is Asternic out there which I’ve used some time ago, but if I recall, it only works with call queues; in this scenario, hunt groups are being used.

Would Asternic meet our needs here or is there another tool I should be looking at?

Thanks!

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@Bradbpw wrote:

We have a remote location where the phones were working fine but stopped recently. I’m having trouble getting all 3 of them back online. I was able to get one Digium D40 phone to register and work as expected at the remote office, however, another Digium D40 and a Sangoma S500 seem to connect to the PBX but not register.

On the D40, the phone appears to connect to the PBX but when dialing an internal extension or external number there is just silence and no connection. The phone also does not appear as a Chan_SIP peer on the PBX. If I change one small thing in the SIP settings (change one digit of the password, extension number, port, or IP address to be incorrect) the call will fail and I get a fail tone when trying to place a call. I have reset the phone to factory and re-entered the config, that didn’t solve it.

On the Sangoma S500, I reset the phone to factory settings then had it auto-provision. The phone found the config and even downloaded our company logo. But, the extension in the upper right had the red line through it and the phone would not place a call. It also would show an “unknown” status on the chan_SIP peers report on the PBX server.

I have the IP for the remote office in the whitelist on System Admin>Intrusion Detection (and restarted intrusion detection) and in the Trusted Zone on the firewall. I have also confirmed that it is not a blocked host on the firewall and not banned on intrusion detection. I event disabled intrusion detection and the firewall while attempting to connect the phone. Neither worked. I also rebooted the PBX server twice and the phones countless times. Mind you, there is a D40 phone at this remote office that connects and works as expected.

I then brought the phones back to the office where the PBX server is. I used my cell phone hot spot and an ASUS RT-N66U router in repeater mode to create an external network separate from the PBX server. I plugged the phones into this external network and they connected and worked perfectly. I did not change any settings between them not working at the remote office and them working on the Hot Spot external network.

I’m at a loss as to why these phones will work on my external hot spot network but not the remote office network, yet there is another phone at the remote office that works. Any ideas where I should start looking?

Thanks,

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@JayArr wrote:

I need a little help with the specific definition of a CID, is it the number only or does it include the text description? Can I route based on a name?

I’ve got a nuisance robocall at 2:20PM every day, it’s one of those fake Google calls. My CDR reports list the CID as:

"Alabama "<12054248547>
"V21513513400050 "<12722000310>
"V21414513700050 "<12722000310>
"California "<13109297077>
"Washington "<13607982701>
"Florida "<13218630551>
"V21114134700042 "<13512050293>
"Alabama "<12054802864>
etc.

The robocall uses a different spoofed number everyday, I can/do blacklist it but since they don’t seem to reuse the number they have become smarter than a simple blacklist.

I’d like to filter all calls that start with a capital V to an IVR that asks the caller to press 1 to continue but so far I’ve had no luck. I tried creating an inbound route with no DID and a CID of _V21. but the calls still get through.

This makes me think the beginning part V21513513400050 that is listed in the CDR report isn’t really part of the CID.

Any direction on what to read to understand this better would be greatly appreciated.

JayArr

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@NLEBob wrote:

Greetings,

I have a PHP script running on a box with Asterisk 11 and FreePBX that works like this:
When a person clicks a button on a special (internal) webpage, the PBX dials the user’s extension; then when they pick up, it dials the number from the phone list that they selected on the web page.
All works fine, except I need the calls to be recorded. I’ve got all the recording checkboxes checked for the extension, but still Asterisk is not recording the call. If I just dial out normally with the phone, it records fine.

My script opens a socket to Asterisk, and sends the following (example):

Action: Originate
Channel: SIP/101
Exten: 18004444444
CallerID: 5852222222
Context: from-internal
Priority: 1
Async: yes
Action: Monitor

What must I do to enable these calls to be recorded?

I did notice a very similar question that was unanswered in the past:
/t/recording-outbound-calls-made-via-originate-context-from-internal/39511

Surely there must be a way to accomplish this. Thanks!

Bob

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@PitzKey wrote:

Hello all,

Early this morning i started the upgrading process by taking a backup first.
Then i used the module 12 to 13 upgrade tool all checks were green upon checking requirements and started the upgrade process, it shortly got stuck with an error:

Stage 1
Bumping Paund PBX to version 13…Done
Checking online servers…Done
Downloading 13 Framework…Done
ERROR: Try running this manually on the CLI to finish: ‘amportal && fwconsole ma upgradeall’

When i ran amportal && fwconsole ma upgradeall it says that this is not a good command and gave the available options for amportal commands.

I then closed the window and figured to maybe try again… now i see this after checking requirements.

image.png978x344 11.2 KB

In system overview i see this:

Currently the PBX is working, thats some good news…

So my question is: whats the proper way to upgrade this machine?

Any help appreciated.
Thanks

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@Waitanchrg wrote:

Hi there!
Make Originate through AMI
"Channel" => “Local/” . $extension . “@from-internal”,
“Exten” => $phoneNumber,
“Context” => “from-internal”,
“Priority” => 1,

After the answer, the channel is swapped
[2017-12-20 10:03:01] VERBOSE[30305][C-00011e35] bridge_channel.c: Channel IAX2/tt86-12931 joined ‘simple_bridge’ basic-bridge
[2017-12-20 10:03:01] VERBOSE[30153][C-00011e35] bridge_channel.c: Channel Local/4873@from-internal-00003bb9;1 joined ‘simple_bridge’ basic-bridge
[2017-12-20 10:03:01] VERBOSE[30153][C-00011e35] bridge_channel.c: Channel IAX2/tt86-12931 left ‘simple_bridge’ basic-bridge
[2017-12-20 10:03:01] VERBOSE[30153][C-00011e35] bridge_channel.c: Channel Local/4873@from-internal-00003bb9;2 left ‘simple_bridge’ basic-bridge <173d60bd-620c-4272-9355-7d5686bfd932>

[2017-12-20 10:03:01] VERBOSE[30153][C-00011e35] bridge_channel.c: Channel IAX2/tt86-12931 swapped with Local/4873@from-internal-00003bb9;2 into ‘simple_bridge’ basic-bridge <173d60bd-620c-4272-9355-7d5686bfd932>

Why is this happening? Which setting affects this?

In this case, on another machine with the same configuration, this does not happen

[2017-12-20 12:01:42] VERBOSE[53211][C-00000044] bridge_channel.c: Channel IAX2/tt86-9882 joined ‘simple_bridge’ basic-bridge <37f8cfa8-e299-459c-95e1-5c6e6d97d0a9>
[2017-12-20 12:01:42] VERBOSE[53161][C-00000044] bridge_channel.c: Channel Local/4801@from-internal-00000033;1 joined ‘simple_bridge’ basic-bridge <37f8cfa8-e299-459c-95e1-5c6e6d97d0a9>
[2017-12-20 12:01:45] VERBOSE[53211][C-00000044] bridge_channel.c: Channel IAX2/tt86-9882 left ‘simple_bridge’ basic-bridge <37f8cfa8-e299-459c-95e1-5c6e6d97d0a9>
[2017-12-20 12:01:45] VERBOSE[53161][C-00000044] bridge_channel.c: Channel Local/4801@from-internal-00000033;1 left ‘simple_bridge’ basic-bridge <37f8cfa8-e299-459c-95e1-5c6e6d97d0a9>

Config:
FreePBX DISTRO
SNG7-FPBX-64bit-1710-1

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@die7fox wrote:

Running FreePBX 14.0.1.20 on a Dell PowerEdge server. We are configuring Polycom VVX400 and VVX600 phones for use on this server. The phones provision fine if we factory reset them from the device and manually enter the following information:

Provisioning server: 10.1.x.x
Protocol: TFTP
DHCP: Custom+Option66
Boot Server Option: 160

Unfortunately, that seems to be the only way to do it. When I enter these settings via the phone’s web interface and reboot it, it always reboots twice and picks up the old provisioning server settings on the second boot. This is a pity, because we have about 90 phones to re-provision and we would prefer to do it remotely. Is there something special we need to do with these Polycom phones to get them to pick up the provisioning server without physically entering the necessary settings?

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@SamTaylor510 wrote:

Just upgraded available online modules from Module Admin. Running Asterisk 14.6.1. Any help is appreciated.

Now receiving the following errors:

Call to undefined method Ucp::getConfig()

/var/www/html/admin/modules/ucp/htdocs/modules/Home/Home.class.php
$feeds = array();
$fpbxfeeds = str_replace("\r","",$fpbxfeeds);
foreach(explode("\n",$fpbxfeeds) as $k => $f) {
$feeds[‘feed-’.$k] = $f;
}
if(!empty($feed) && !empty($feeds[$feed])) {
$feeds = array($feeds[$feed]);
}
$widgets = array();
$reader = new Reader;

    //Check if dashboard is installed and enabled,
    //if so then we will use the same cache engine dashboard uses
    if($this->UCP->FreePBX->Modules->moduleHasMethod("dashboard","getConfig")) {
        $storage = $this->UCP->FreePBX->Dashboard;
    } else {
        $storage = $this->UCP->FreePBX->Ucp;
    }
    foreach($feeds as $k => $feed) {
        $etag = $storage->getConfig($feed, "etag");
        $last_modified = $storage->getConfig($feed, "last_modified");
        $content = '';
        try {
            $resource = $reader->download(trim($feed), $last_modified, $etag);
            if ($resource->isModified()) {
                $parser = $reader->getParser(
                    $resource->getUrl(),
                    $resource->getContent(),
                    $resource->getEncoding()
                );

                $content = $parser->execute();
                $etag = $resource->getEtag();
                $last_modified = $resource->getLastModified();

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@mitterhuemer wrote:

Hello,

i made 2 fresh FreePBX SNG7 1712 installs today.

Both distros have the same problem now.

After i upgrades all modules (edge mode) and enabling a LetsEncrypt Cert with certman i cannot enable this cert anywhere.

HTTPS Setup tells me:

To set up this server for SSL (HTTPS) access you will need to either install or update your version of Certificate Manager from Module Admin

And in the SIP Settings i cannot set a default cert for TLS Connections.

But Certman 13.0.36.11 is installed.

I already tried to remove and reinstall the certman module but nothing helped. FreePBX tells mi i should install certman.

My default cert is installed

image.png1794x85 6.71 KB

I cannot select a certificate to install.

image.png1758x295 20.3 KB

I have some more distros running certman 13.0.36.11 without this problem. But they where installed months berfore and just got a upgrade from time to time.

Can someone help me solving this? My first idea was reinstalling the distro…but i already setup 2 machines today with the same error from the iso…

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@Ruteng wrote:

Installed free pbx after burning 4 dvds. Now it asks for login:
I made a password during install but no prompt for login name. Now my pc is a brick. Pls help.

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@rgriffinefd wrote:

Installing a new system and seem to everything working with the exception of being able to pick up parked calls. I can transfer calls without any problems and can park the calls, but when trying to pick them up it hangs up on the call.

FreePBX 13.0.1.192.19
Asterisk 13.0.7.1
Preloaded Asterisk Modules

Module
Action
pbx_config.so
chan_local.so
func_db.so
res_odbc.so
res_config_odbc.so
cdr_adaptive_odbc.so

Manually Loaded Modules

Module
Action
format_wav.so
format_pcm.so
format_mp3.so
res_musiconhold.so
res_parking.so

Excluded Modules

chan_woomera.so
pbx_gtkconsole.so
pbx_kdeconsole.so
app_intercom.so
chan_modem.so
chan_modem_bestdata.so
chan_modem_i4l.so
app_trunkisavail.so
chan_alsa.so
chan_oss.so
app_directory_odbcstorage.so
app_voicemail_odbcstorage.so
chan_modem_aopen.so
chan_woomera.so
cdr_radius.so
cel_radius.so
cdr_mysql.so
res_phoneprov.so
res_config_ldap.so
res_config_sqlite3.so
res_clialiases.so
chan_mgcp.so
cdr_custom.so
app_minivm.so
cel_custom.so

Using SIPSTATION and have installed endpoint manager as well. Disabled Parking Pro but didn’t make a difference. Have a Peplink Balance 20 router hoping someone can point me in the right direction.

[2017-12-20 17:54:58] WARNING[19105][C-00000013] channel.c: Unable to find a codec translation path: (gsm|g722|alaw|ulaw) -> (g729)
[2017-12-20 17:54:58] WARNING[19105][C-00000013] file.c: Unable to open beep (format (g729)): No such file or directory
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] bridge_channel.c: Channel SIP/1501-00000029 joined ‘simple_bridge’ basic-bridge <8fb45003-7190-4e23-b6cc-575d06099660>
[2017-12-20 17:54:58] WARNING[19105][C-00000013] channel.c: No path to translate from SIP/1501-00000029 to SIP/fpbx-1-wURXKekK0UDd-00000026
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] bridge_channel.c: Channel SIP/1501-00000029 left ‘simple_bridge’ basic-bridge <8fb45003-7190-4e23-b6cc-575d06099660>
[2017-12-20 17:54:58] VERBOSE[9340][C-00000012] file.c: <SIP/fpbx-1-wURXKekK0UDd-00000026> Playing ‘beep.ulaw’ (language ‘en’)
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] app_macro.c: Spawn extension (macro-parked-call, s, 22) exited non-zero on ‘SIP/1501-00000029’ in macro ‘parked-call’
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Spawn extension (from-internal, 11, 1) exited non-zero on ‘SIP/1501-00000029’
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Executing [h@from-internal:1] Macro(“SIP/1501-00000029”, “hangupcall”) in new stack
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/1501-00000029”, “1?theend”) in new stack
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/1501-00000029”, “0?Set(CDR(recordingfile)=)”) in new stack
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/1501-00000029”, “”) in new stack
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/1501-00000029’ in macro ‘hangupcall’
[2017-12-20 17:54:58] VERBOSE[19105][C-00000013] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1501-00000029’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] bridge_channel.c: Channel SIP/fpbx-1-wURXKekK0UDd-00000026 left ‘simple_bridge’ basic-bridge <8fb45003-7190-4e23-b6cc-575d06099660>
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] app_macro.c: Spawn extension (macro-dial, s, 22) exited non-zero on ‘SIP/fpbx-1-wURXKekK0UDd-00000026’ in macro ‘dial’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Spawn extension (ext-group, 1599, 14) exited non-zero on ‘SIP/fpbx-1-wURXKekK0UDd-00000026’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [h@ext-group:1] Macro(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “hangupcall,”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “1?theend”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “0?Set(CDR(recordingfile)=)”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/fpbx-1-wURXKekK0UDd-00000026’ in macro ‘hangupcall’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Spawn extension (ext-group, h, 1) exited non-zero on ‘SIP/fpbx-1-wURXKekK0UDd-00000026’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] app_stack.c: SIP/fpbx-1-wURXKekK0UDd-00000026 Internal Gosub(crm-hangup,s,1) start
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:1] NoOp(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “Sending Hangup to CRM”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:2] NoOp(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “HANGUP CAUSE: 16”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:3] ExecIf(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:4] NoOp(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “MASTER CHANNEL: 1513810486.84 = 1513810486.84”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:5] GotoIf(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “0?return”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:6] Set(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “__CRM_HANGUP=1”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:7] AGI(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “sangomacrm.agi”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2017-12-20 17:54:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] res_agi.c: <SIP/fpbx-1-wURXKekK0UDd-00000026>AGI Script sangomacrm.agi completed, returning 0
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] pbx.c: Executing [s@crm-hangup:8] Return(“SIP/fpbx-1-wURXKekK0UDd-00000026”, “”) in new stack
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] app_stack.c: Spawn extension (ext-group, h, 1) exited non-zero on ‘SIP/fpbx-1-wURXKekK0UDd-00000026’
[2017-12-20 17:54:59] VERBOSE[9340][C-00000012] app_stack.c: SIP/fpbx-1-wURXKekK0UDd-00000026 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2017-12-20 17:55:02] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5078’ (callid: 76338444d6afe6a15b48b1d97c8ffead) - No matching endpoint found
[2017-12-20 17:55:03] VERBOSE[5347] asterisk.c: Remote UNIX connection
[2017-12-20 17:55:03] VERBOSE[23060] asterisk.c: Remote UNIX connection disconnected
[2017-12-20 17:55:03] VERBOSE[5347] asterisk.c: Remote UNIX connection
[2017-12-20 17:55:03] VERBOSE[23082] asterisk.c: Remote UNIX connection disconnected
[2017-12-20 17:55:03] VERBOSE[5347] asterisk.c: Remote UNIX connection
[2017-12-20 17:55:03] VERBOSE[23100] asterisk.c: Remote UNIX connection disconnected
[2017-12-20 17:55:29] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:55:46] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5086’ (callid: 29d975e5860a5bfd2e959c5d8f5619bf) - No matching endpoint found
[2017-12-20 17:55:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:56:28] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5094’ (callid: 129f50faa927e3017d97c1d1362b54d9) - No matching endpoint found
[2017-12-20 17:56:29] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:56:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:57:11] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5080’ (callid: db95a32d38681efdbdaac41240060a76) - No matching endpoint found
[2017-12-20 17:57:29] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:57:53] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5090’ (callid: 0d733d12c83fff8933f782705c25f0be) - No matching endpoint found
[2017-12-20 17:57:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:58:29] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:58:36] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5077’ (callid: b32795be76c88147531629f9d8758cd3) - No matching endpoint found
[2017-12-20 17:58:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:59:18] NOTICE[18986] res_pjsip/pjsip_distributor.c: Request ‘INVITE’ from ‘“2006” sip:2006@173.246.206.69’ failed for ‘104.217.195.50:5090’ (callid: 1f6a277c31064b81079770841222d1f4) - No matching endpoint found
[2017-12-20 17:59:29] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500
[2017-12-20 17:59:59] NOTICE[7319] chan_sip.c: Received SIP subscribe for peer without mailbox: 1500

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