vicidial.org

Just finished our initial testing of Asterisk 16 support and everything is working as far as I can tell. Thankfully Digium did not make any major changes to the Asterisk Manager Events that Vicidial uses so everything pretty much just worked. No configuration files need to be changed since Asterisk 13. It has not been tested on a production system as of yet, but it has placed tens of millions of calls on my dev system without a crash. You can download a fully patched version of the Asterisk source code here:

http://download.vicidial.com/beta-apps/ ... ici.tar.gz

Please keep in mind that support is still in beta. Use this at your own risk and do not cry to us if it crashes. If you are not sure how to compile and install Asterisk from scratch for use with Vicidial this is not something I recommend messing with. Also keep in mind that App MeetMe is required by Vicidial but is not compiled by default so you will need to enable that.

One completely optional feature I did add was IAX and SIP peer event logging. This logs when SIP and IAX peers Register, Unregister, Go Reachable, Go Unreachable, Go Lagged, Disconnect due to lack of RTP, and Experience a Critical Timeout to the vicidial_peer_event_log. This should allow us in the future to notify agents and managers when peers are having connection issues. To activate this you need to be running an SVN version from 2021-04-08 (Matt is working on the code commit at the time of me writing this). And add this to the /etc/asterisk/manager.conf file in the [listencron] section:

; Optional
eventfilter=Event: PeerStatus
eventfilter=Event: SIPRTPDisconnect
eventfilter=Event: PeerRegistered
eventfilter=Event: SIPCriticalTimeout

Matt may add this to the config when he commits the code. I am going to be adding a patch for Asterisk 13 to enable this feature in that version as well.

thank you sir, im actually testing 16 now but via the vicidial opensuse repo which is the 16.16.2 version. Will this version work as well? I seem to be having some issues with dahdi(meetme)

-- Called 55558600052@default
[Apr 7 14:30:02] WARNING[9418][C-00000006]: pbx.c:2927 pbx_extension_helper: No application 'MeetMeAdmin' for extension (default, 55558600052, 1)
[Apr 7 14:30:02] == Spawn extension (default, 55558600052, 1) exited non-zero on 'Local/55558600052@default-00000002;2'
[Apr 7 14:30:02] WARNING[9418][C-00000006]: pbx.c:2927 pbx_extension_helper: No application 'AGI' for extension (default,

I have not tested that package yet. Kumba was setting up an initial build of that package for me to test once the source code release was done. I am not sure if it even compiles app_meetme.so yet.

One thing I did forget to mention is that you will need to enable the res_speech.so module in modules.conf to get the res_agi.so module to load. Not sure why res_agi.so depends on that now but that would explain the last error you were getting.

ok thank you

here is an install instructions to do thw switch for anyone else interested: https://cyburityllc.com/?p=1913

I hope this helps

-Nox

Ok, got this working on Centos7 scratch install.

Question, when running the install.pl script, we need to put 16.X, 16.17.0-vici, or leave at 13.X?

Also in the server settings, leave at 13 or change to 16.17.0-vici?

I put the actual version in both places. 16.17.0-vici

Is Sangoma CPA still supported in Asterisk 13 and 16?
Can't see any patches related to it applied or available, or are these patches no longer required?

Sangoma CPA/Lyra/ParaXip went end-of-life in 2017 and is no longer sold or supported by Sangoma. As such, we have not spent time to add it's functionality to newer versions of Asterisk.

Hello

Whats the difference between https://downloads.asterisk.org/pub/tele ... ent.tar.gz
and http://download.vicidial.com/beta-apps/ ... ici.tar.gz

or asterisk-13-current.tar.gz
and http://download.vicidial.com/beta-apps/ ... ici.tar.gz

will it work if I download asterisk from downloads.asterisk.org?

no, there is changes in the -vici versions that make it work correctly with the astguiclient....

thanks , I hope theres a guide somewhere on how to patch asterisk's version

Hi
I followed scratch install centos8

https://cyburityllc.com/?p=2033&related ... position=0

but I'm getting
"Your session has been disabled
CLICK HERE TO RESET YOUR SESSION"

I've been trying to find solutions at the forums for a day now but none of them work, can anyone help?

also whenever I modify 6666 user after install it does not update the database, I have to manually update 6666 copy the SQL log @ ADMIN CHANGE LOG at the database

yea my instructions were flawed, i missed the mariadb installation on the steps. apt-get install mariadb-client mariadb-server

@carpenox
this is centos 8 scratch w/ asterisk 16 guide you made

I did yum install mariadb mariadb-server

but still getting "Your session has been disabled", and modifying user on the admin portal is not persisting at the database

please start a new thread for that and ill be happy to help you, but i dont want to overpower this post with technical issues other than in direct relation to Vicidials supported versions such as those on Leap....

How is asterisk 16 looking so far compared to 13 in terms of stability and performance?

The AMD is definitely better, stability has been on par as well.

@mcargile Thank you for this info. Very helpful. We are having fun with Asterisk 13 AMD sending 50% more answering machine calls to agents on a new ViciBox 9.0.1 installation. Looking forward to v 16 when part of a standard ViciBox install/upgrade.

What I wanted to pickup here is your SIP peer event logging addition. We're getting this error:
DBD::mysql::db do failed: Unknown column 'peer_status' in 'field list' at /usr/share/astguiclient/AST_manager_listen_AMI2.pl line 693.

It seems that we're missing the SQL update scripts for adding the peer_status & ping_time columns to the phones table for that this update works:
UPDATE phones set peer_status = '$event_hash{'PeerStatus'}', ping_time = '$event_hash{'PingTime'}' where server_ip='$server_ip' and extension='$phone_extension' and protocol='$event_hash{'ChannelType'}';";

extras/upgrade_2.14.sql has the "CREATE TABLE vicidial_peer_event_log", but I've checked right up to SVN version 3456 and can't find any "ALTER TABLE phones ADD peer_status & ping_time ..."

--
ViciBox 9.0.1 | Leap 15.2 | Version: 2.14-814a | BUILD: 210609-2108 | SVN Version: 3452 | DB Schema Version: 1615 | Asterisk 13.29.2-vici

Looks like I forgot to send Matt the alter statements. Just sent those to him. He should get them added to SVN soon.

Nice, good catch Roger

Those two DB fields are now added in the most recent svn/trunk revision 3458.

Thanks!

Testing out AST16, have installed on both Vicibox install and a Centos scratch install.

The issue that I am seeing on both boxes is that the UPDATECRON is always running, and havent seen that with AST13. And, on the server report, the times are always 2-3 seconds different. In the astguiclient.conf and in the server settings I have tried both asterisk versions 13 and 16.

SERVER - DESCRIPTION IP ACT LOAD CHAN AGNT DISK TIME VER
TESTast Test install of Asterisk server 192.168.0.100 Y / Y / Y 7 - 1% 0 0 54% 2021-06-20 20:51:52 3461
PHP Time 2021-06-20 20:51:55
DB Time 2021-06-20 20:51:55


== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1
== Manager 'updatecron' logged off from 127.0.0.1
== Manager 'updatecron' logged on from 127.0.0.1

Can you please connect to the ASTupdate screen session using this command:

Code: Select all

screen -rx ASTupdate

and post the output here. To disconnect from it, press CTRL-A followed by the D key.

Did another reinstall. Centos8 scratch with Asterisk 16.17.0-vici

Installed with SVN 3463.

After installation and reboot, changed the Asterisk Version to 13.38. Rebuild config files and submit. Asterisk reloads, but still showing the UPDATECRON. Tried the RELOAD from SYSTEM SETTINGS and same. Ran the command and below is the output.

Again rebuilt config in SERVER SETTINGs still same. Rebooted again, and all is normal. Seems the rebuild script is missing something, and only updates on a reboot.

ERROR: 2021-06-23 18:08:34|67|command timed-out|
ERROR: 2021-06-23 18:08:38|68|command timed-out|
ERROR: 2021-06-23 18:08:41|69|command timed-out|
ERROR: 2021-06-23 18:08:44|70|command timed-out|
ERROR: 2021-06-23 18:08:47|71|command timed-out|
ERROR: 2021-06-23 18:08:50|72|command timed-out|
ERROR: 2021-06-23 18:08:54|73|command timed-out|
ERROR: 2021-06-23 18:08:57|74|command timed-out|
ERROR: 2021-06-23 18:09:00|75|command timed-out|

You can put Asterisk 16 as the version. The code handles that just fine. What does your /etc/asterisk/manager.conf file look like?

Put 16.X in astguiclient.conf and in Admin > Servers correct?

and manager.conf is default install

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
timestampevents = yes

[cron]
secret = 1234
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate

[updatecron]
secret = 1234
read = command,reporting
write = command,reporting

eventfilter=Event: CoreShowChannel


[listencron]
secret = 1234
read = system,call,log,verbose,command,agent,user,dtmf
write = command

eventfilter=Event: Shutdown
eventfilter=Event: DTMFBegin
eventfilter=Event: DTMFEnd
eventfilter=Event: NewCallerid
eventfilter=Event: Newstate
eventfilter=Event: Hangup
eventfilter=!Event: HangupRequest

; Optional
eventfilter=Event: PeerStatus
eventfilter=Event: SIPRTPDisconnect
eventfilter=Event: PeerRegistered
eventfilter=Event: SIPCriticalTimeout

[sendcron]
secret = 1234
read = command
write = system,call,log,verbose,command,agent,user,originate

[sipcron]
secret = 1234
read = call
write = command

eventfilter=Event: SIPInvite
eventfilter=Event: SIPResponse

That all looks correct. Try connecting to the AMI interface:

Code: Select all

#> telnet localhost 5038

Then copy and paste this in:

Code: Select all

Action: login
Username: updatecron
Secret: 1234


Then press enter twice. If that succeeds run this:

Code: Select all

Action: CoreShowChannels

Then hit enter twice. Copy the results and paste it here.

This server has AST16 with the astguiclient.conf set to 16.X, and the Admin > Servers asterisk version set as 16.17.0-vici

[root@sip2 ~]# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/5.0.3
Action: login
Username: updatecron
Secret: 1234

Response: Success
Message: Authentication accepted

Action: CoreShowChannels

Response: Error
Message: Permission denied



This server has AST16, but astguiclient.conf set to 13.X and Admin > Servers asterisk version 13.38.2-vici

[root@sip1 ~]# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/5.0.3
Action: login
Username: updatecron
Secret: 1234

Response: Success
Message: Authentication accepted

Action: CoreShowChannels

Response: Success
EventList: start
Message: Channels will follow

Event: CoreShowChannelsComplete
EventList: Complete
ListItems: 0




On a AST13 box, when choosing Rebuild Conf Files=Y, and Musiconhold=Y, the reload shows in ast cli
dialer*CLI>
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Reloading module 'res_musiconhold.so' (Music On Hold Resource)
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
dialer*CLI>


In the AST16 box, does not show anything but sendcron

And what I have noticed is the only way to rebuild the conf files is to reboot. This 3rd box with AST16 in both astguiclient.conf, and in Admin > Servers......... if I change both to AST13 and reboot, the telnet and sendcron will be normal.

Steps:
Complete install, and reboot after install.pl
Log into fresh install and give permissions to Admin user
In Admin > Servers change asterisk version to 13.38.2-vici, Rebuild conf files=Y. (no change, still show updatecron and no reload in asterisk)
Reboot a 2nd time, and I am assuming on the reboot the conf files are being rebuilt. (asterisk cli is normal, phones connect and make calls)

When you telnet to localhost port 5038 you are directly connecting into the Asterisk Manager Interface. It has absolutely nothing to do with any settings in Vicidial. Astguiclient.conf is not read. Admin -> Servers is not accessed. As such if both servers are running Asterisk 16 with identical manager.conf config files then those commands should work flawlessly independent of your Vicidial / astguiclient.conf settings. You need to compare the two manager.conf files and make sure everything is set correctly.

As for rebuilding config, that just sounds like you are not properly starting Asterisk in a screen session using the Vicidial start_asterisk_boot.pl script. Instead it sounds like the Asterisk init script is starting Asterisk. As such Vicidial is unable to connect to Asterisk screen session and issue commands which is the only way to reliably reload config files without crashing Asterisk.

So to verify that there isnt an issue I installed Vicibox 10 Beta, installed Asterisk 16, set the Asterisk version to 16.16.2 and rebooted. Everything came up correctly. The update script connected once to Asterisk and has not disconnected.

Is this beta support in production already? Asterisk 13 EOL was in October 2021.
Is there a compiled package with Asterisk 16 already in vicidial repositories, or we still need to compile it from source?

We do not have any active clients using Asterisk 16 to the best of my knowledge. That said you can run the 'vicibox-ast16' command on Vicibox 10 to install it. I have not personally run this script. You would have to ask Kumba in the Vicibox section of this forum more about that script.

We do have a few premise clients using Asterisk 16 in production at this point, all report it to be stable and have been using it for several months.

I stand corrected.

Hey mflorell and mcargile,

Looking to install this patched Asterisk build and had a few questions;

- Is the tarball attached the latest version or is there a newer one?
- Do we have any further updates from the premise clients on their experience (good or bad) on the stability of 16 and ViCiDial.
- From which SVN revision did the database schema changes come in. Apologies, I haven't followed this that much. We've been sorta happy on 13.
- What tests do you recommend when using the new version? My go tos were the usual run of the mill stuff; auto-dialling, manual dialling, inbound dialling, maybe a little bit of load testing with a steady stream of concurrent calls.

Thanks guys,
Martin.

That is the latest tarball version we've released, but I believe we have put out a more recent OpenSuSE package of Asterisk 16.

Premise clients, and now VICIhost clients also, have all had positive experiences with Asterisk 16. No issues reported to date.

If you're using Asterisk 13, you don't have to upgrade VICIdial for Asterisk 16 compatibility. If you want to use some of the new optional Asterisk 16 features, you will need svn/trunk revision 3422.

As for tests, yes, just logging in as an agent to the agent screen and placing auto-dial and manual calls, as well as taking inbound calls.

mflorell wrote:That is the latest tarball version we've released, but I believe we have put out a more recent OpenSuSE package of Asterisk 16.

Premise clients, and now VICIhost clients also, have all had positive experiences with Asterisk 16. No issues reported to date.

If you're using Asterisk 13, you don't have to upgrade VICIdial for Asterisk 16 compatibility. If you want to use some of the new optional Asterisk 16 features, you will need svn/trunk revision 3422.

As for tests, yes, just logging in as an agent to the agent screen and placing auto-dial and manual calls, as well as taking inbound calls.

Thanks for the support on this, Matt. I'll get testing and report any issues I found.

Cheers,
Martin.

any quick way to install this like before when we upgraded from asterisk 11 to asterisk 13 with a command?

install-asterisk16 or something??? every time i have tried installing via packages I have issues.

-kashinc

I totally missed the vicibox-ast16... my bad.