vicidial.org

We just posted 2.2.0 RC 8 to sourceforge

https://sourceforge.net/projects/astguiclient/files/


Please test it out if you can and we should be able to do a full release soon.

There are many more small bug fixes as well as minor feature enhancements in this RC, and at this point 2.2.0 is installed on several dozen of our client sites and is proven to be very stable.

If you are upgrading from rc7 or earlier please be sure you add the following to extensions.conf:

exten => _8320*.,1,AGI(VD_amd.agi,${EXTEN}-----YES)
exten => _8320*.,2,Hangup


Here is a list of the rest of the changes since 2.0.5:

http://www.vicidial.org/VICIDIALforum/v ... php?t=9854

Very well with this version.
All function can run well. Ready upgrade all server to this version.
But I cannot understand with the in-group on hold time setting,
When I set `Hold Time Option Seconds` = '30' and `Hold Time Option` = 'PRESS_VMAIL ', but when call in ,the VMAIL voice not sound at 30 second but almost seem random.

Checked the code, and find the hold time value is calculated from others.
But some one can give me advice to how to set the hold time to a suit value. And our old flow is callmenu-> music on hold(20s)->on hold prompt -> music on hold(20s) -> hold option messgaes(can press 1 to Vmail)... Thanks.

Surrounding:
centos 2.6.18-92.e15.vnow
Astersik 1.2.30.2
vicidial VERSION: 2.2.0-236 BUILD: 100413-2328

Hold Time Option settings are only activated when the ESTIMATED hold time is greater than the hold time seconds value.

Oh, I see. That's great. Thank you mflorell.

I noticed that the listen and barge links are missing in the 2.2.0 RC8 version. Is this intentional?

yes, we removed those so people would use the built-in monitoring of the real-time screen instead. The links are just commented out in the HTML if you need them for some reason.

I downloaded and ran perl install.pl, but when I look in system settings I still see;

Version: 2.2.0rc7
DB Schema Version: 1193
DB Schema Update Date: 2010-03-30 05:00:15


what did I miss?

logandzwon wrote:...what did I miss?

I bet that's reading the UPGRADE document

There are no functional database updates from rc7 to rc8, so you can skip that step.

okli wrote:

logandzwon wrote:...what did I miss?

I bet that's reading the UPGRADE document

Thanks guys,
UPGRADE document didn't call out different instructions to go from 2.2.0 beta to 2.2.0, it gave 2.0.5 to 2.2.0. Looking at the .sql included for 2.0.5 to 2.2.0 it looks like there are not any sql changes from b7 to b8.


So far beta 8 looks good, except the agents HATE the new sound that plays when they are connected to a customer. That sound is making them angry!!!

What sound does it play when it connects the customer?

all you need to do to upgrade to a newer RC is to run the perl ./install.pl script

mflorell wrote:What sound does it play when it connects the customer?

all you need to do to upgrade to a newer RC is to run the perl ./install.pl script

When it adds someone to the meetme it plays a sound to the conference to let them know someone has been added to the meeting. For whatever reason, between updated from (vici2.2.0b7 and asterisk1.4.27) to (vici2.2.0b8 and 1.4.27.1) that sound changed and my agents hate it.

What does the sound sound like?

mflorell wrote:What does the sound sound like?

It sounds like a stabbing double chime. Before updating it sounded like bal-oop.

You need to either use the ViciDial Group patched Asterisk or apply the changed enter and leave sounds and then recompile your Asterisk.

http://download.vicidial.com/required-a ... ici.tar.gz

or

cd ../asterisk-1.4.21.2
wget http://www.eflo.net/files/enter.h
wget http://www.eflo.net/files/leave.h
mv -f enter.h apps/enter.h
mv -f leave.h apps/leave.h

mflorell wrote:You need to either use the ViciDial Group patched Asterisk or apply the changed enter and leave sounds and then recompile your Asterisk.

http://download.vicidial.com/required-a ... ici.tar.gz

or

cd ../asterisk-1.4.21.2
wget http://www.eflo.net/files/enter.h
wget http://www.eflo.net/files/leave.h
mv -f enter.h apps/enter.h
mv -f leave.h apps/leave.h

Looks like the vici enter/leave sounds got left out out of the 1.4.27.1-vici.

root@VICI:~# ls -la /usr/src/asterisk-1.4.27-vici/apps/enter.h
-rw-r--r-- 1 root root 6174 2009-02-18 14:16 /usr/src/asterisk-1.4.27-vici/apps/enter.h
root@VICI:~# ls -la /usr/src/asterisk-1.4.27.1-vici/apps/enter.h
-rw-r--r-- 1 root root 16995 2006-02-14 14:14 /usr/src/asterisk-1.4.27.1-vici/apps/enter.h


root@VICI:/usr/src/asterisk-1.4.27.1-vici# head apps/enter.h
/*
* U-law 8-bit audio data
*
* Source: enter.raw
*
* Copyright (C) 1999, Mark Spencer and Linux Support Services
*
* Distributed under the terms of the GNU General Public License
*
*/

That is a possibility, I have not personally installed that one. We will look into it.

Yep. I forgot to put those in. I have uploaded a fixed version to http://download.vicidial.com

mcargile wrote:Yep. I forgot to put those in. I have uploaded a fixed version to http://download.vicidial.com

Should we use 1.4.30 or 1.4.27-1?

Thanks.

1.4.30 has not been tested by us at all. It just has the patches applied to it. If you are feeling daring feel free to use it and let us know if you have any problems.

mflorell wrote:yes, we removed those so people would use the built-in monitoring of the real-time screen instead. The links are just commented out in the HTML if you need them for some reason.

Hello Mr. Matt,

I have started using 2.2.0.rc8 but didnt find SIP Barge & Listen. As you mention "people would use the built-in monitoring of the real-time screen instead. " what is that ? Will i be able to do bargin with that ? if YES than HOW ??

Thanks

Yes, you can barge and listen, you need to add a phone entry(if one does not exist) for the phone you want to monitor from, then put that phone login into the real-time screen and select "MONITOR" from the pull-down menu, then you can monitor. This works with any kind of phone, not just softphones like the old method. Also, it works across multiple servers.

There is a tutorial that covers this in the manager manual.

After selecting monitor & giving user name i found LISTEN coming near to session id. But when i click on LISTEN nothing is happening. I have register softphone with one of the user on the system...

You need to use a PHONE LOGIN, not a username in the monitor field of the realtime screen.

ekaftan wrote:

mcargile wrote:Yep. I forgot to put those in. I have uploaded a fixed version to http://download.vicidial.com

It seems the tar.gz you uploaded was not packed correctly i get:

Code: Select all

# tar -xvvvvzf asterisk-1.4.27.1-vici.tar.gz
gzip: stdin: not in gzip format
tar: Child returned status 1
tar: Error exit delayed from previous errors


Thanks!

Alejandro

does vicidial group uses

asterisk-1.4.27-vici.tar.gz

or

asterisk-1.4.21.2-vici.tar.gz ?


which one is more stable? and worry free on dtmf?

We use both, depending on the installed distro and whether we need DAHDI or zaptel.

As for DTMF, there really doesn't seem to be much difference in our experience between the two.

I should mention that we have solved DTMF issues by using Sangoma cards that have the Octasic chips in them(HWEC T1/E1 cards and the D100 transcoding cards) which can take the DTMF signal processing on the card and do a much better job at it than Asterisk does.