vicidial.org

I usually don't comment much on pending development, but this one is going to require database software from 2012 or newer, either MySQL 5.6+ or MariaDB 5.3+. That's when "microseconds" support was added in each. Current VICIdial code requires only MySQL 5.1+(from 2005), and I know a lot of premise clients that are using very old versions like this.

This will be for some new SIP logging features that will hopefully help to be able to better identify FAS(False Answer Supervision) and PDD(Post Dial Delay).

Any comments?

https://dev.mysql.com/doc/refman/5.6/en ... conds.html
https://mariadb.com/kb/en/library/micro ... n-mariadb/

Anyone running a system that old (and sticking to it fiercely) should be able to put their DB on a separate server running a newer version of mysql. Unless there's a challenge with the mysql-client version, that should allow legacy systems to continue to upgrade without re-customizing their systems (which is the usual reason for refusal to upgrade). IMHO.

Yes, this shouldn't be a deal-breaker for most clients, but I know of several of our clients that are running older databases, which is one of the main reasons I posted this. The new features were actually committed to svn/trunk on Friday and are running in production for a couple of our clients already.

Are these logs optional and off by default so there won't be any crashes until the new feature is activated?

Yes, these new features are optional, are disabled by default and have to be enabled in several places before the logging will occur.

The results so far are fascinating, and we haven't even written any reporting for it yet. One example is a phone call placed by a client today that went from INVITE to 200-OK in 0.177043 seconds(without any ringing), which should be fairly impossible and is almost certainly FAS.

Useful tool. Until the FASsholes find out about it and learn to mimic a real call better. Although it's likely that most of them will never bother. So very cool. I expect a client to request an "alert" system eventually for when they begin to crop up.

I am trying to enable this on my test system and having an issue. SIP_EVENT_LOGGING.txt says to patch asterisk 13 with the patch in the in the extras folder. I have tried to apply it via patch sip_logging.patch. it seems to be looking for a directory that doesn't exist. it seems like it is looking for asterisk 13.21.1 and i am running the current one of 13.27.0. do i need to update the patch to apply it? if so what would the current dir of the asterisk that it is trying to use be at?

You often have to alter the path to fit your own directory structure, we have tested the patch on 13.17 and it does apply cleanly, so give it a try.

ok, thank you i will see if i can track down the asterisk path that it is calling for. this seems like a great addition.

Hello, How can I disable this on manual dialing?

Are you saying you want this to be active for auto-dialing but disabled for all manual dial calls through the agent screen?

mflorell wrote:Are you saying you want this to be active for auto-dialing but disabled for all manual dial calls through the agent screen?

yes, is this possible

Yes, I just went ahead and added that last night, so you'll need to upgrade to the latest svn/trunk revision.

The changes in how you configure it in the Settings Container are detailed in the doc,
http://vicidial.org/docs/SIP_EVENT_LOGGING.txt

mflorell wrote:Yes, I just went ahead and added that last night, so you'll need to upgrade to the latest svn/trunk revision.

The changes in how you configure it in the Settings Container are detailed in the doc,
http://vicidial.org/docs/SIP_EVENT_LOGGING.txt

Nice! Which server will I update? Thank you

Well, I don't know what servers you have so I can't say

When you upgrade to a more recent svn/trunk revision, you should always upgrade all servers in a cluster, never just one.

mflorell wrote:Well, I don't know what servers you have so I can't say

When you upgrade to a more recent svn/trunk revision, you should always upgrade all servers in a cluster, never just one.

Haha, I see, Thank you. I tried on my express server first, i tried to autodial 200 calls and 10 manual dial, it doesn't show the records on sip event logging but it shows total number of records @_@.

is there a write up on how to patch the asterisk?