@Chuckak wrote:

Current PBX Version:14.0.13.4
Current System Version:12.7.6-1904-1.sng7

If I leave the browser, Chrome, Firefox or Edge on the FreePBX dashboard for more than a few hours the Network widget quits updating. Then awhile later I cannot navigate to any tabs. I cannot refresh the browser.
Closing and reopening sometimes works but usually not. Can’t find webpage. If I change browsers it will work for awhile again. If I use the IP address instead of FQDN it will work until it times out again.

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@moussa854 wrote:

FYI, I would like to share these two articles hoping to encourage the community in keeping their system up to date.

In summary, attackers could use a vulnerability to access the FreePBX, steal data and install crypto mining scripts. It seems that the security hole has been patched, so it is recommended not to put off updates for long.

Source: https://www.virusbulletin.com/conference/vb2019/abstracts/asterisk-targeted-voipspionage-campaign/

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@FreerPBXer wrote:

I’m reposting this from a locked thread because it didn’t get an answer.

I’ve read both these articles, just to preface.

https://wiki.freepbx.org/display/FPG/Time+Groups+Sample+Configurations
https://wiki.freepbx.org/display/FPG/Time+Group+User+Guide

In order to ease holiday scheduling we switched to defining non-working hours. And the two articles don’t say what happens if the start time for a time condition is after the end time (i.e. start at 5:00 PM and end at 8:00 AM). And if the end day for such a condition is Thursday, does it end with the change from positive match to negative at 8:00 AM, or from negative to positive at 5:00 PM? Similarly, with Monday would it start with a neg match at 8:00 AM or with a positive at 17:00?

If we need to define separately as 00:00 to 08:00 and also 17:00 to 23:59, then it seems like there would be a one minute unmatched state every night from 23:59 to 00:00.

Lastly, what if there is a conflict? i.e. we could define 08:00 to 17:00 M-F as work hours and set call flow based on that. But if we define a Monday as a holiday and designate after-hours call flow, which group ‘wins’?

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@tomas12343 wrote:

Hello everyone. We have a CRM that keeps tracks of the salesmen and each one of them has a specific client list. So, I want the CRM to write in a database of freepbx the CID’s, the extension of the user (salesman) and the client’s name and freepbx to transfer the inbound calls (based on CID) to the specific extensions from the database plus show the name of the client on the extension. We can handle the CRM part, but unfortunately I don’t know how to code in the freepbx (only UI stuff). Does anyone have any idea how to do this?

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@dstu wrote:

Hello,

I have an old FreePBX v2.11.0.43 (asterisk version 11.25.2) at one end (sys1) and a FreePBX v14.0.13.6 (asterisk version 13.20.0) in the other (sys2), using PJSIP on port 5060. Both are connected behind firewalls and have static public IP addresses (external IP and local network are properly configured on both ends).

I’ve been struggling for many hours with trying to make calls between them.

I was trying initially to create trunks without authentication, but sys1 kept rejecting the calls coming from sys2:

SIP Peer ACL: Rejecting ‘xx.xx.xx.xx’ due to a failure to pass ACL ‘(BASELINE)’

Then, I tried to add a username and secret to the trunk (on both ends), but that didn’t solve the problem either. I also disabled in sys2’s trunk pjsip advanced settings the field “Permanent Auth Rejection”, but that also didn’t solve the problem.

Can anyone suggest trunk settings for both ends to enable site to site communications?

Thank you very much in advance for your kind assistance.

David

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@VoIPTek wrote:

Since yesterday several instances are losing communications ( not sure they are failed or crashed ) and in hunting through logs I’m seeing some strange firewall errors, first below is messages info, and with there is also this line, looking at several areas without a solid reason for the lockup

HP Warning: Wrong license type, license codes are not matching this host or license text has been altered (license file: /etc/schmooze/schmooze.zl). in /usr/lib/sysadmin/licensed.php on line 0
PHP Warning: License check failed! in /usr/lib/sysadmin/licensed.php on line 0
Starting firewall.
1570920605: Wall: 'Firewall service now starting.

Oct 12 15:39:46 bpbx php: Wall: 'Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012' returned 0

Oct 12 15:40:33 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:41:21 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:42:08 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:42:55 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:43:42 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:44:29 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:45:16 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:46:04 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:46:51 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:47:39 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:48:26 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:49:13 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:50:00 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:50:47 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:51:34 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:52:21 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:53:09 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:53:56 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:54:43 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:55:31 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:56:18 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:57:05 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:57:52 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:58:39 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 15:59:26 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:01:07 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:01:56 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:02:43 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:03:31 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:04:19 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:05:06 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:05:54 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:06:41 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:07:29 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:08:16 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:09:04 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:09:51 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:10:40 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:11:27 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:12:14 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:13:02 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0
Oct 12 19:13:49 bpbx php: Wall: ‘Firewall Rules corrupted! Restarting in 5 seconds#012More information available in /tmp/firewall.log#012’ returned 0

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@psdk wrote:

Hi to all,

We are using last distro with latest updates with two Sangoma cards; A102DE and A400DE. In the GUI, we have the digital card in the list and we can configure it, but analog card is not detected!
Cards are OK and we can see in lspci or dahdi_hardware and we have tested with other Asterisk system.

So is there any problem with new updates and version 14?

Regards

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@Josh9591 wrote:

Hello again.

So I recently had an established IP address of 192.168.2.12 No connection problems what so ever. I had a bug issue where I’d drop calls every 30 second due to a static ip not being set.

I’ve made a reset to the system several times to obtain an ip address of my local Xfinity network. And now, it looks like when I reboot the machine, I see a 50/50 chance of FreePBX not being able to Obtain an IP Address when FreePBX is launched. When I had a connection with freepbx I Could connect to it using Ssh and the GUI just fine. When I go to the system modules. the Page times out. And now i’m unable to make an established ip address to connect to the freepbx using gui. The last error in the console was “unable to write /etc/wanpipe/global.conf disable sangoma DIGIUM or Change Permissions to /etc/wanpipe/global.conf” witch I belive caused to drop connections.

when I did “# fwconsole chown” it re wrote the permissions. But then that was like it.

fwconsole chown

I have FreePBX installed on a eMachine Computer I found running through a cisco 24 port unmanaged switch. and I’m on XFinity comcast internet. Thanks Guys!!

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@ITconsultant wrote:

on my dashboard it says Xmpp is Not running
XMPP Version 14.0.1.19
uninstalling and reinstalling from GUI not resolving it
when I do a fwconsole restart I get the following error

[root@nypbx ~]# fwconsole restart
Running FreePBX shutdown…

Core FastAGI Server is not running
Stopping RestApps Server
Stopped RestApps Server
Stopping UCP Node Server
[>---------------------------] < 1 sec
Stopped UCP Node Server
Chat Server is not running
Stopping Zulu Server
Stopped Zulu Server
Shutting down Asterisk Gracefully. Will forcefully kill after 30 seconds.
Press C to Cancel
Press N to shut down NOW
[============================] 3 secs
Wanrouter: No valid Sangoma Hardware found, if you have no Sangoma cards this is OK
Stopping DAHDi for Digium Cards
DAHDi Stopped
Queue Callback Server is not running
Queue Callback Event Server is not running
Running FreePBX startup…
Running Asterisk pre from Dahdiconfig module
Wanrouter: No valid Sangoma Hardware found, if you have no Sangoma cards this is OK
Starting DAHDi for Digium Cards
DAHDi Started
Running Asterisk pre from Firewall module
Running Asterisk pre from Sysadmin module
Running Sysadmin Hooks
Restarting fail2ban
fail2ban Restarted
Updating License Information for 19971868
Checking Vpn server
Starting Asterisk…
[============================] 7 secs
Asterisk Started
Running Asterisk post from Core module
Running Asterisk post from Dahdiconfig module
Running Asterisk post from Endpoint module
Running Asterisk post from Pagingpro module
Running Asterisk post from Restapps module
Starting RestApps Server…

Broadcast message from root@nypbx.cf (Mon Oct 14 21:56:42 2019):

Firewall service now starting.

[>---------------------------] < 1 sec
Started RestApps Server. PID is 5886
Running Asterisk post from Ucp module
Starting UCP Node Server…
[>---------------------------] 1 sec
Started UCP Node Server. PID is 6222
Running Asterisk post from Vqplus module
Queues Pro is not licensed.
Running Asterisk post from Xmpp module
[>---------------------------] < 1 secResetting PBX Users Failed: The command “node /var/www/html/admin/modules/xmpp/node/resetpbxusers.js” failed.

Exit Code: 1(General error)

Working directory: /root

Output:

Error Output:

/var/www/html/admin/modules/xmpp/node/node_modules/mongodb/lib/mongo_client.js:421
throw err
^
MongoError: failed to connect to server [localhost:27017] on first connect [MongoError: connect ECONNREFUSED 127.0.0.1:27017]
at Pool. (/var/www/html/admin/modules/xmpp/node/node_modules/mongodb-core/lib/topologies/server.js:336:35)
at emitOne (events.js:116:13)
at Pool.emit (events.js:211:7)
at Connection. (/var/www/html/admin/modules/xmpp/node/node_modules/mongodb-core/lib/connection/pool.js:280:12)
at Object.onceWrapper (events.js:317:30)
at emitTwo (events.js:126:13)
at Connection.emit (events.js:214:7)
at Socket. (/var/www/html/admin/modules/xmpp/node/node_modules/mongodb-core/lib/connection/connection.js:189:49)
at Object.onceWrapper (events.js:315:30)
at emitOne (events.js:116:13)
at Socket.emit (events.js:211:7)
at emitErrorNT (internal/streams/destroy.js:64:8)
at _combinedTickCallback (internal/process/next_tick.js:138:11)
at process._tickCallback (internal/process/next_tick.js:180:9)

Running Asterisk post from Zulu module
Starting Zulu Server…
[>---------------------------] 1 sec
Started Zulu Server. PID is 6581
[root@nypbx ~]#

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@thepossum905 wrote:

I have a fairly new freepbx 14 installation, the phones are on a remote network from the hosted pbx. Generally things are running swimmingly, although we’ve recently just enabled g729 in order to save on bandwidth.

There are some intermittent network issues with the internet provider, causing occasional call quality issues. All the boring stuff yup, a bit of extra packet loss and jitter that we can’t predict or prevent.
So we need to somehow visualize the per-call quality, in some sort of automated manner.

From the asterisk CLI, I can manually “pjsip show channelstats” and see live info on current active calls. Or I can script a “asterisk -rx ‘pjsip show channelstats’” to see everything at a periodic interval.

But what would be much more helpful would be for asterisk to log the per-call stats at the end of each call.

My google fu repeatedly brings up a command “rtcp set stats on”, but … nothing changes in the logs. How can I see the produced jitter data so that I can start to visualize it?

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@hgcrane wrote:

I am having trouble reaching rs.sangoma.net from a Salt Lake City IP address. This issue is causing interruptions in my customer’s phone service. After many traceroutes, I believe this to be a routing issue at CyberLynk in Wisconsin, but they won’t talk to me because I’m not a customer. This issue has been going on for over a week. I know people at Sangoma read these posts. I’m hoping Tony Lewis or someone else at Sangoma will take an interest in this and tell me where to go with it. I have a ticket open with Sangoma. Ticket number 931883.

PLEASE HELP

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@voipgenius wrote:

So…are we finally going to get SMS integration?

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@FreerPBXer wrote:

I’m reposting this from a locked thread because it didn’t get an answer.

I’ve read both these articles, just to preface.

https://wiki.freepbx.org/display/FPG/Time+Groups+Sample+Configurations
https://wiki.freepbx.org/display/FPG/Time+Group+User+Guide

In order to ease holiday scheduling we switched to defining non-working hours. And the two articles don’t say what happens if the start time for a time condition is after the end time (i.e. start at 5:00 PM and end at 8:00 AM). And if the end day for such a condition is Thursday, does it end with the change from positive match to negative at 8:00 AM, or from negative to positive at 5:00 PM? Similarly, with Monday would it start with a neg match at 8:00 AM or with a positive at 17:00?

If we need to define separately as 00:00 to 08:00 and also 17:00 to 23:59, then it seems like there would be a one minute unmatched state every night from 23:59 to 00:00.

Lastly, what if there is a conflict? i.e. we could define 08:00 to 17:00 M-F as work hours and set call flow based on that. But if we define a Monday as a holiday and designate after-hours call flow, which group ‘wins’?

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@emrettt wrote:

[2019-10-19 11:54:52] ERROR[8980] pjproject: sip_transport.c Error processing 420 bytes packet from UDP 185.37.24.71:5060 : PJSIP syntax error exception when parsing ‘Via’ header on line 2 col 34:
SIP/2.0 200 Keepalive
Via: SIP/2.0/UDP 192.168.1.9:5160:5160;rport=23842;branch=z9hG4bKPj95d08c5f-9367-45a9-aafc-f95703f79245;received=94.123.100.173
From: <sip:hello@192.168.1.9>;tag=e9c8ee7f-38a4-4761-b4b9-3af1dc7ad799
To: <sip:185.37.24.71>;tag=89757bbff68d7d0abdeaa0f6fc52d389.06aa
Call-ID: 126a39c5-e6a6-4f02-8e6b-7615a9272bed
CSeq: 13177 OPTIONS
Server: kamailio (4.3.7 (x86_64/linux))
Content-Length: 0

I made some reseach but couldnt find any solution.why I getting this

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