Your session has been disabled | webphone | rocky 8

[dgxa]

Hello, I just installed Vicidial on Rocky 8, and when I configure WebPhone and log in, it doesn't allow me to connect to the session, indicating "Your session has been disabled."

When I connect to the softphone, the same thing happens, even with firewall rules allowed.

I suspect the vicidial_live_agents table isn't recording the agent login, but it isn't corrupted.

cat /var/www/html/agc/vicidial_mysqli_errors.txt
2025-03-29 21:05:01|vdc_db_query|00135|


There are screens on:
2708.ASTVDremote (Detached)
2693.ASTVDauto (Detached)
2673.ASTlisten (Detached)
2661.ASTsend (Detached)
2658.ASTupdate (Detached)
2153.asterisk (Detached)
2144.astshell20250329191312 (Detached)
2134.ASTconf3way (Detached)
2121.ASTfastlog (Detached)
2108.ASTVDadapt (Detached)
10 Sockets in /run/screen/S-root.

screen -r 2658
|cid_channels_recent_010000012001|SHOW TABLES LIKE "cid_channels_recent_010000012001";
----- AMI Version 5.0.5 -----


SVN Version: 3836 || Version: 2.14b0.5 || mysql Ver 15.1 Distrib 10.3.39-MariaDB || Rocky Linux 8.10 (Green Obsidian) || Asterisk 16.30.1-vici

[carpenox]

there could be many reasons, im guessing dahdi isnt running, try typing "dahdi_cfg -v"

If that doesnt work, you can try the other steps on this debug article:

https://dialer.one/index.php/how-to-deb ... -vicidial/

Chris

[dgxa]

there could be many reasons, im guessing dahdi isnt running, try typing "dahdi_cfg -v"

If that doesnt work, you can try the other steps on this debug article:

https://dialer.one/index.php/how-to-deb ... -vicidial/

Chris

Dahdi is running, the funny thing is that it even happens with the softphone

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[root@vmi2531555 ~]# dahdi_cfg -v
DAHDI Tools Version - 3.4.0

DAHDI Version: 3.4.0
Echo Canceller(s):
Configuration
======================


0 channels to configure.

Code: Select all

iptables -L -nv
Chain INPUT (policy DROP 364K packets, 68M bytes)
 pkts bytes target     prot opt in     out     source               destination
27214 2902K ACCEPT     all  --  lo     *       0.0.0.0/0            0.0.0.0/0
 9657  804K ACCEPT     all  --  *      *       0.0.0.0/0            0.0.0.0/0            state RELATED,ESTABLISHED
    4   208 ACCEPT     tcp  --  *      *       0.0.0.0/0            0.0.0.0/0            tcp dpt:23022
  148  8564 ACCEPT     tcp  --  *      *       0.0.0.0/0            0.0.0.0/0            tcp dpt:22
    0     0 ACCEPT     all  --  *      *       10.b.c.d/16          10.b.c.d/16
    4   208 ACCEPT     tcp  --  *      *       a.b.c.d          0.0.0.0/0            tcp dpt:80
  104  5408 ACCEPT     tcp  --  *      *       a.b.c.d          0.0.0.0/0            tcp dpt:443
    0     0 ACCEPT     tcp  --  *      *       a.b.c.d          0.0.0.0/0            tcp dpt:5060
    1   583 ACCEPT     udp  --  *      *       a.b.c.d          0.0.0.0/0            udp dpt:5060
   41 14744 ACCEPT     udp  --  *      *       0.0.0.0/0            0.0.0.0/0            udp dpts:10000:55000

vmi2531555*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
gs102/gs102               a.b.c.d                              D  Yes        Yes            45567    OK (228 ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0 offline]



[carpenox]

you shouldnt post your IP's here, there is bad actors hiding in the depths, however try with the firewall off altogether to see if thats the issue

[williamconley]

Hello, I just installed Vicidial on Rocky 8, and when I configure WebPhone and log in, it doesn't allow me to connect to the session, indicating "Your session has been disabled."

When I connect to the softphone, the same thing happens, even with firewall rules allowed.

If soft phone doesn't work, webrtc phone isn't worth attempting.

"does not allow me to connect to the session" and "session has been disabled" are interesting observations, but please remember to indicate where you see/get this information and post that detail.

Onward! Baby Steps:

Will the soft phone register?

If the soft phone will register: Will the soft phone ring when the agent attempts login? (Asterisk CLI output during the login attempt might be useful if it does not ring)

If the soft phone will not register: Asterisk CLI output during a registration attempt. Verify account/extension and secret. It may be necessary to turn on SIP debugging in asterisk if there's no output to find out what's really going on.

Are these agent soft phones local or somehere else on the internet? Does the server have a public IP and/or a Private IP?

[dgxa]

you shouldnt post your IP's here, there is bad actors hiding in the depths, however try with the firewall off altogether to see if thats the issue

That was not the real IP address but I change the IP how you suggest.

Firewall was disabled for this issue to avoid firewall interruptions

[carpenox]

and was it still not worknig with firewall off?

[dgxa]

and was it still not worknig with firewall off?

Firewall is opened(disable)
The softphone registered
The softphone not ringing when the agent attempts login

I found this error when I try to connect
https://drive.google.com/file/d/1TylWkjvVruC8j0CBVRloJP8hSfZ1fdV2/view?usp=sharing

[williamconley]

Can the soft phone dial out?

Can you manually dial the soft phone from another phone after it has registered?

Post the soft phone sip account information and admin->phones settings (without password of course).

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asterisk -rx "sip show peers"

Just the output for this phone

[carpenox]

its very possible your ISP is blocknig the ports, what area are you in? alot of the middle east ISP's have the ports blocked recently

[dgxa]

Can the soft phone dial out?

Can you manually dial the soft phone from another phone after it has registered?

Post the soft phone sip account information and admin->phones settings (without password of course).

Code: Select all

asterisk -rx "sip show peers"

Just the output for this phone

Code: Select all

vmi2531555*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
100/100                   x.y.v.z                              D  Yes        Yes            39054    OK (226 ms)
200/200                   x.y.v.z                               D  Yes        Yes            26602    OK (127 ms)
gs102/gs102               (Unspecified)                            D  Yes        Yes            0        UNKNOWN


Call Between extensions successful

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Connected to Asterisk 16.30.1-vici currently running on vmi2531555 (pid = 1966)
  == Using SIP RTP CoS mark 5
       > 0x7f1900021410 -- Strict RTP learning after remote address set to:  x.y.v.z :17854
    -- Executing [200@default:1] Dial("SIP/100-0000000c", "SIP/200,60,") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/200
    -- SIP/200-0000000d is ringing
    -- Dial Tiemout Reset on channel SIP/200-0000000d due to RINGING message
       > 0x7f190400a570 -- Strict RTP learning after remote address set to:  x.y.v.z :18168
    -- SIP/200-0000000d answered SIP/100-0000000c
    -- Channel SIP/200-0000000d joined 'simple_bridge' basic-bridge <33515a5c-6da6-4eb8-813d-dbd04626819b>
    -- Channel SIP/100-0000000c joined 'simple_bridge' basic-bridge <33515a5c-6da6-4eb8-813d-dbd04626819b>
       > 0x7f190400a570 -- Strict RTP qualifying stream type: audio
       > 0x7f190400a570 -- Strict RTP switching source address to  x.y.v.z :18164
       > 0x7f1900021410 -- Strict RTP qualifying stream type: audio
       > 0x7f1900021410 -- Strict RTP switching source address to  x.y.v.z :18191
       > 0x7f1900021410 -- Strict RTP learning complete - Locking on source address  x.y.v.z :18191
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
       > 0x7f190400a570 -- Strict RTP learning complete - Locking on source address  x.y.v.z :18164
    -- Channel SIP/200-0000000d left 'simple_bridge' basic-bridge <33515a5c-6da6-4eb8-813d-dbd04626819b>
    -- Channel SIP/100-0000000c left 'simple_bridge' basic-bridge <33515a5c-6da6-4eb8-813d-dbd04626819b>
  == Spawn extension (default, 200, 1) exited non-zero on 'SIP/100-0000000c'
    -- Executing [h@default:1] AGI("SIP/100-0000000c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----8-----SIP 200 OK)") in new stack
    -- <SIP/100-0000000c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----8-----SIP 200 OK) completed, returning 0
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
vmi2531555*CLI>


[root@vmi2531555 ~]# systemctl status firewalld
Unit firewalld.service could not be found.
[root@vmi2531555 ~]#
[root@vmi2531555 ~]#
[root@vmi2531555 ~]# iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source destination

Chain FORWARD (policy ACCEPT)
target prot opt source destination

Chain OUTPUT (policy ACCEPT)
target prot opt source destination
[root@vmi2531555 ~]#

I starting to suspect that I have a missing dependency or something like that

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[carpenox]

you can attempt to recompile asterisk

[williamconley]

Can the soft phone dial out?

Can you manually dial the soft phone from another phone after it has registered?

You reposted my questions. You did not answer them, but you did repost them. LOL