Congestion Issues

[Deepak365]

my setup
agents = 6-8
dial level = 16-20
hopper level = 1000
goautodial = 2.4
asterisk = 1.4.39(tried to upgrade to 1.8 but still showing 1.4)
pc = i3,4gb,1tb

dialplan=
exten => _1XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1XXXXXXXXXX,2,Dial(SIP/${EXTEN}@goautodial,,tTo)
exten => _1XXXXXXXXXX,3,Hangup


disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g726
type=friend
username=xxxxxx
fromuser=xxxxx
secret=xxxxx
host=00.000.00.000
fromdomain=00.000.00.000
dtmfmode=rfc2833
context=trunkinbound

problem is even having level of 16-20 calls are coming very slow whereas my voip balance is getting exhausted. i checked all my calls most of the calls are of 3-4 seconds which didnt come to my agents. i talked with my voip provider, he confirmed that your half of the balance is going as a waste but he said there is no issue with voip. so i went to asterisk cli and watched the log while calls were ringing. most of the calls were congestion,circuit is busy,congestion,cancel,thats it.
its effecting my work a lot. need serious help

[bobchaos]

VoiP balance? I'd consider a different provider o.O There's a lot of billing models out there, most of which are more forgiving of dialers. Dialers on pay-per-use systems will murder your bill!

That being said, it sounds to me like you're dialing more lines than you have. In Admin --> Server you can tell Vicidial how many lines a given server is connected to so that it will know it's limitations. You can also adjust how many lines are available on a per campaign basis. It also sounds like you may be working a bad list, hence all the short calls. How is Vicidial coding those calls?

Also, I'm not sure what you mean exactly by "tried to upgrade to 1.8 but still showing 1.4", but that sounds like problems waiting to happen. I'd consider a fresh install with up to date software. Finally,, as far as I know, there is nowhere in the world where both alaw and ulaw are in use. You should see which is in use by your provider and match it to minimize the negotiation phase of your SIP calls.

[Deepak365]

I changed my VoIP provider after posting the previous post but same issues. My VoIP provider changed the lines and trunk in admin from default to 900. I was checking my calls in asterisk via putty. I saw there congestion ,circuit is busy,everyone is busy,got sip response 503. Another thing which was a lot was playing sip silence. I removed the alaw from account entry. please help to make my call flow better

[williamconley]

1) Tried to upgrade to 1.4 ... You may well have broken something. Is this an old system you've taken over?

2) Please ALWAYS post your Vicidial Version with Build. Goautodial version 2.4 does not exist. That may be your vicidial version. If so, i's Very Old and you should look at upgrading. Note that goautodial is merely an installer for Vicidial. You may want to consider re-installing it (to fix any broken asterisk issues you may have caused) or you may want to consider installing with Vicibox.com's .iso image instead (Later version of Vicidial, more toys, more recent code with more bug fixes ... and nothing broken by the modified Goautodial interface changes). Vicibox installs STOCK Vicidial, which is the most tested version across the world.

3) You have looked at logs and seen congestion ... what percentage of the calls are congested? You did not post any asterisk CLI output from any of your problem calls.

4) What you appear to be saying is "I have lots of problems ... please fix all of them", but you've posted no details about any of them. You may have observed some of them in the Asterisk CLI, but you did not share any of that with the rest of us. So all we can really do is make vague suggestions that may have no relevance to your situation.

My best advice would be a fresh server with fresh Vicidial from Vicibox and a clean link to at least three Carriers to be sure you can easily resolve any Congestion issues by switching carriers (or by using all three to avoid any limitations on channels).

Happy Hunting!

[Deepak365]

Thanks for replying. Now issue has been solved. One thing i want to know as you said use all three of them, can i use multiple carrier for a single campaign?

[williamconley]

You can, if you set it up to do so in a special carrier.

It's possible to send XX calls through the first, XX through the second and the rest through the third, for instance. Or even to choose the one with the least live calls now and send through that one.

Not "Standard", but there have been a few posts with code that works for this purpose.

[Deepak365]

Ok but william i dont have that kind of knowledge like yours so it seems difficult to me. One more thing is when i download a used/called list (lets say around 10k numbers) i see that status of around 7000 numbers is NA , VDAD. I want to know that why dialer throws away 70% of list(any) in NA

[williamconley]

Which is why google is your friend. You'll have to do the work of finding those entries (no time tonight, sorry).

And your assertion that "dialer throws away" is interesting. If you look up the System Statuses, you'll see that NA = No Answer Autodial. Which means there was no answer. The dialer is doing its job by reporting that it did not get an answer when dialing that number. I'm not sure it's appropriate to characterize this as being the fault of the dialer.

There may be underlying problems causing those No Answer results: Such as exceeding the channel limits at your carrier or even improperly formatted phone numbers. But that in no way means that the dialer is "arbitrarily killing perfectly good phone numbers". LOL

[Deepak365]

Ok! Got it. Thanks for explaining this .