vicidial.org

Hello everybody !

Let me tell you, my Server story in a few words.
I installed ViciBox v6.0.2 for the First Time at 17-09-2014,as a Vicibox Express installation.
After that i have used this server without making new upgrades, till 2 months ago (i was doing daily ADMIN_backup).
My HDD got broken so i needed to change it and make a new installation.
I made the installation with the same CD as first time.As it finished ive seen a new vicidial version(with a new logo and 1 button for pause and resume at the agent interface).
After that i restored the backup the Vicidial logo changed to the old one and other options too (during the DB restore ive seen some query's with Warnings).

Now my server settings are like this:
Version : 2.10b0.5
SVN Version : 2153
DB Schema Version :1385
DB Schema Update Date : 2014-09-17

I want to know what update do i have to make to upgrade to the last version ?
do i have to use upgrade_2.10.sql or upgrade_2.12.sql?

i have tryed using upgrade_2.10.sql but after ive done that my SVN Version turned to 0

From the information you have give, here are the steps I think you should do:
- cd /usr/src/astguiclient/trunk
- svn update
- perl install.pl
- mysql asterisk
- \. /usr/src/astguiclient/trunk/extras/upgrade_2.10.sql
- \. /usr/src/astguiclient/trunk/extras/upgrade_2.12.sql
- quit
- shutdown -r 0

Thanks for your answer Matt.So the idea is to update me SVN folder use install.pl and update the db with the 2 scripts.
Im doing this because i want to add 2 telephony servers .
Is this upgrade required to add the addtional telephony servers ?
I will use the same CD with Vicibox-install

An upgrade is recommended, but not required. If you don't perform an upgrade you do need to make sure that all servers on your cluster are operating on the same version of the software.

About the procedure that you said.i did it and now it seems OK.
First i made everystep and tried to access vicidial,but it shows no users no campaign ,nothin at all.

After ive restarted the server,everything turned to be OK.

An upgrade is recommended, but not required. If you don't perform an upgrade you do need to make sure that all servers on your cluster are operating on the same version of the software.

OK now that i have upgraded the All-in-One Server,i think i will be good to go,by installing the telephony servers with Vicibox-Install right ?
Thanks

Glad to hear everything works now

Yes, you should be good to go to add more servers to your cluster now.

Hello
After updating the All-In-One Server ,2 weeks later i added dhe 2 additional telephony servers (with vicibox-install).
Server are like this :
1. 192.168.1.200 All-in-one server (asterisk 1.8.29.0-vici,SVN 2366)
2. 192.168.1.201 Telephony 1 (asterisk 1.8.32.3-vici , SVN 2366)
3. 192.168.1.202 Telephony 2 (asterisk 1.8.32.3-vici , SVN 2366)

I want to know are there any additional changes i have to make to the servers?
I changes my existing phones to the new servers.
What about the carrier ? How to make the new telephony server to dial ?
i have one carrier which can be registred on different servers and dial.How to use this carrier on all servers ?

Thank You

Under admin->Carriers, copy the existing carrier and use the IP of each of the new servers. you'll need a unique carrier ID for each of the servers, of course, but otherwise the new entries should be identical to the original IF you are not using registration.

If you are using registration: Avoid using the same registration entry on two or more servers. Try to get a unique account for registration of each server.

Also look at the bulk phone adder (in the admin utilities) and look into using phone aliases. If your SIP phones can register to more than one account (3CX, Zoiper), you can configure phone aliases (easy with the bulk adder, just avoid any phone numbers you already have) and you'll have load balancing automatically as your agents log in and out all day.

Happy Hunting

Thanks for your reply williamconley
I have done what you said and it dials at all the servers.
Now im having a problem with the calls ,like if an agent is incall for 30 minutes the customer voice disapper.so the agent have to hang up and call the customer again.
how is it possible ?
Thanks

You'll need to check the call logs and the asterisk cli to find out why the call terminated.

Is it always 30 minutes? Or is that an estimate? Is it all agents or some? Is it all servers or one of them? Are the calls on one server or is the agent on one server?

Note that the Real Time Screen has options to allow you to view the Client Call Server (for the client portion of the phone call) as well as the Agent Call Server so you can determine which server seems to have the problem.

this happens at all the operators who are 30 minutes in Call.yes this happens at all servers 1.201 and 1.202.call and agents are on both servers.
i havent seen this problem before adding the 2 telephony servers.

williamconley wrote:You'll need to check the call logs and the asterisk cli to find out why the call terminated.

You'll need to do this for one of the calls. And try a call from a phone without the Vicidial interface and be sure it can last more than 30 minutes. Could be your carrier killing the call.

If i go to User Stats of a specific agent who have this problem i see the call with this problem that have a Hangup : CALLER.
The same thing to the other agents who have the same problem.

I this information enough ?

Ill try dialing out of the Vicidial Interface and see what happens.

You'll need the Asterisk CLI logs. Most likely.

Calling without the vicidial interface may be helpful. If that call has the same problem it'll be a Carrier setting under Admin->Carriers or a problem with your Telephone company. Or both.

Making a call outside of the Vicidial Interface that lasts more than 30 minutes will be difficult.(as i see you are suggesting me to test my Carrier and my SIP provider,RIGHT ?)

I have 2 Different SIP Accounts from different providers.Today i will user only the second SIP provider and see what happens.
If this still happens i thinks it will be a Vicidial or a Carrier configuration problem .
Am i right ?

Thanks

Hello
i have made the tests.
as i explained i have 2 sip accounts :
1.TRING (this is the one who does not let a call more than 30 minutes)
2.BIBI

With BIBI i dont have the disconnection 30 minutes call problem.
The TRING staff says that they dont have problems from their part.

Im posting my carrier settings.Because maybe its something in there

TRING Carrier Account Entry:
[tring]
type=friend
username=XXXXXXXXXX
secret=XXXXXXXXXX
host=XXXXXXXXXX
dtmfmode=rfc2833
nat=yes
directmedia=no
qualify=yes
context=default
insecure=port,invite
fromuser=XXXXXXXXXX
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=g723

BIBI Carrier Account Entry:

[bibi]
fromuser=XXXXXXXXXX
username=XXXXXXXXXX
user=XXXXXXXXXX
fromdomain=XXXXXXXXXX
type=peer
qualify=yes
secret=XXXXXXXXXX
port=5070
nat=yes
insecure=very
host=XXXXXXXXXX
canreinvite=no
disallow=all
allow=g729

while i was dialing with TRING i made a ngrep of number that passed 30 minutes and this was the hangup script:
U 192.168.1.201:5060 -> 95.107.130.50:5060
BYE sip:00390396956879@95.107.130.50:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 95.107.162.230:5060;branch=z9hG4bK7456382a;rport.
Max-Forwards: 70.
From: "V8271010190004652610" <sip:32616161616@95.107.162.230>;tag=as4ddc1d9d.
To: <sip:00390396956879@95.107.130.50>;tag=SDtovm599-45826b5f-13c4-55dec5eb-adc68f43-33fe.
Call-ID: 53f7667b30d4ba8b447d0dbc5aa089b4@95.107.162.230:5060.
CSeq: 104 BYE.
User-Agent: Asterisk PBX 1.8.32.3-vici.
Proxy-Authorization: Digest username="32616161616", realm="default", algorithm=MD5, uri="sip:00390396956879@95.107.130.50:5060", nonce="f82ab54ce39e", response="7ff99361294c0c4c7251b1662c1e6a7a".
X-Asterisk-HangupCause: Unknown.
X-Asterisk-HangupCauseCode: 0.
Content-Length: 0.
.

#
U 95.107.130.50:5060 -> 192.168.1.201:5060
SIP/2.0 481 Unknown Dialog.
Via: SIP/2.0/UDP 95.107.162.230:5060;received=95.107.162.230;branch=z9hG4bK7456382a;rport=1091.
From: "V8271010190004652610" <sip:32616161616@95.107.162.230>;tag=as4ddc1d9d.
To: <sip:00390396956879@95.107.130.50>;tag=SDtovm599-45826b5f-13c4-55dec5eb-adc68f43-33fe.
Call-ID: 53f7667b30d4ba8b447d0dbc5aa089b4@95.107.162.230:5060.
CSeq: 104 BYE.

Okay it just turned out that the SIP provider had problems.

This problem is still happening.After Tring Provider made test it results they were no problems.
Now i made some tests myself.
I added some phones at the old server 1.200 (All-In-One Server).And the interruption of the calls made with Tring SIP after 30 minutes does note happen.
It happens only to the new servers. 1.201(Telephony 1) and 1.202(Telephony 2).

How is this possible ?????
Can it be a Carrier configuration problem ?