vicidial.org

64 Bit Install
VERSION: 2.4-309a BUILD: 110430-1642 1.4.39.1-vici.go RPM by demian@goautodial.com currently running on go (pid = 29331)

I have this under account entry GoAutoDial GUI

[Xcast_OutBound]
type=friend
host=*
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
canreinvite=no
context=trunkoutbound

[Xcast_Inbound]
type=friend
host=*
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
canreinvite=no
context=trunkinbound

[Connecium]
type=friend
host=*
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
canreinvite=no
context=trunkinbound

This is for my dialplan entry goautodial GUI.

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,n,GotoIf($[${GROUP_COUNT(sgw)}<80]?10)
exten => _91XXXXXXXXXX,n,NoOp("Max Trunks Exceeded")
exten => _91XXXXXXXXXX,n,Congestion
exten => _91XXXXXXXXXX,10,Set(GROUP()=sgw)
exten => _91XXXXXXXXXX,n,NoOp("TIU: ${GROUP_COUNT(sgw)}")
exten => _91XXXXXXXXXX,n,Dial(SIP/${EXTEN:1}@Connecium,,To)
exten => _91XXXXXXXXXX,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?20)
exten => _91XXXXXXXXXX,n,Hangup
exten => _91XXXXXXXXXX,20,Dial(SIP/${EXTEN:1}@Xcast_OutBound,,To)
exten => _91XXXXXXXXXX,n,Hangup

I found your example on another post, trying to setup fail over if there is congestion, or a trunk fails.
This is what the stats in the real time report look like for this campaign.

DIAL LEVEL: 25.000 TRUNK SHORT/FILL: 0 / 0 FILTER: INCOME TIME: 2013-02-21 10:33:30
DIALABLE LEADS: 99297 CALLS TODAY: 9625 AVG AGENTS: 6.00 DIAL METHOD: RATIO
HOPPER ( min/auto ): 200 / 0 DROPPED / ANSWERED: 176 / 803 DL DIFF: -0.03 STATUSES: AA, DROP, AB, NEW, B
LEADS IN HOPPER: 590 DROPPED PERCENT: 21.92% DIFF: -0.42% ORDER: DOWN
AGENT AVG WAIT: 15 AVG CUSTTIME: 26 AVG ACW: 6 AVG PAUSE: 9
DROP IN-GROUP STATS - DROP PERCENT: 0% CALLS: DROPS/ANSWERS: /
CARRIER STATS: HANGUP STATUS 24 HOURS 6 HOURS 1 HOUR 15 MIN 5 MIN 1 MIN
ANSWER 20494 5027 3431 622 100 0
BUSY 387 105 69 12 0 0
CANCEL 20189 3428 2341 373 70 0
CHANUNAVAIL 2032 0 0 0 0 0
CONGESTION 4375 773 578 161 13 0
NOANSWER 1 1 1 1 0 0

I understand some of the stats might look a little funky due to testing. Still getting congestion, and when I watch the CLI, I don't seem to be seing any calls transferred over to the other trunk.

Help me Obi Won Kenobi, your my only hope.

I'm not sure where you got this from, but the "groups" samples I put up were for using specific numbers of channels at specific carriers, thus allowing 50 channels from carrier A, then 20 channels from carrier B, then 30 channels from carrier C, totalling 100 channels distributed across three carriers keeping the usage of each limited to the allowed channels for each carrier. It was not designed as a failover.

There are plenty of other failovers which you could try, of course. But I've never posted one specifically designed to work for failover (although I've used many of them ...).

If you mix your failover attempt with a grouping attempt, you'll likely confuse the issue and not be able to get it to work. Failover in asterisk is not complex enough for that!

I am constantly perusing the forums here to try to better educate myself. And while looking for fail over, I found this comment in a post of yours on a forum. The reason why I am looking for fail over, is because when I am running the dialer hard, to keep agent wait time low, sometimes the lines are congested and when they try to make a outbound call they get the famous "Dial timed out error, contact system administrator".

there have been several failover methods posted, and failover is a standard method of operation in Asterisk itself. The question is when you want to activate it ... congestion? you can check for congestion and send the call to another extension number or named function ... in Asterisk Dialplan language. However, many will simply add +101 to the extension and put in another dial, or add +1 to the extension and put in another dial command ... depending on your version of asterisk and settings. as I said, there have been many failover methods and there are many more available in places like asterisk-guru. Just remember that the Dial command that is executed must remain the same one in Vicidial. You can certainly change the carrier through which the call will go, but the dial command options can't change (especially that "o" switch!).

Well I see things like this,


Postby Op3r » Thu Dec 03, 2009 2:59 pm
Yes this can be used as a fail over if the first dial fails.

exten => _91NXXNXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_1.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_2.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_3.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_4.com)
exten => _91NXXNXXXXXXXX,n,Dial(SIP/${EXTEN1}@sbc.provider_5.com)
exten => _91NXXNXXXXXXXX,n,Hangup

And then I hear how it will once for each trunk and then pass through the one that gets through. Is this true?

Or Mr. Conley I also found this.
http://wasdak.blogspot.com/2011/02/vici ... -hunt.html
exten => _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8X.,2,Dial(${TRUNK1}/0${EXTEN:1},,tTor)
exten => _8X.,3,Dial(${TRUNK2}/0${EXTEN:1},,tTor)
exten => _8X.,4,Hangup
exten => _8X.,103,Dial(${TRUNK2}/0${EXTEN:1},,tTor)
exten => _8X.,104,Hangup
exten => _8X.,204,Hangup

Which in my case needs to look something like this....

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${TRUNK1}/0${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,3,Dial(${TRUNK2}/0${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,4,Hangup
exten => _91XXXXXXXXXX,103,Dial(${TRUNK2}/0${EXTEN:1},,tTor)
exten => _91XXXXXXXXXX,104,Hangup
exten => _91XXXXXXXXXX,204,Hangup

Really, I have two outbound trunks, and one inbound trunk. I would like it to dial either one randomly. To even out the load, or go to the other if one is busy. If you could enlighten me on how to do this I would greatly appreciate it.

Trunk Names are as follows.
Xcast_OutBound
Connecium

You could use a trick from another post that uses the last digit of the number being dialed to determine which carrier to attempt. In either case, the "other" carrier would be the failover. You can use a "gotoif" (which is in some of the examples) to send the call to the path chosen.

Thank you for the support William.