Cannot make autodial campaign work

[chrislog]

OK, here is the problem:

I installed the latest vicibox release that runs on ubuntu server. What I want to do is to Set up an auto-dial campaign to dial without any live agents and play a message. I followed all the tutorials from the manager manual and I can make it dial manually, but when I try to have it dial automatically it says that it is unable to create channel of type zap, cause 34. I get the same error when I try to set it as per tutorial F. Here is the CLI:

Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/10452281211859@default-d007,2", "agi://127.0.0.1:457 7/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/10452281211859@default-d007,2", "Zap/g1/04522812118 59||To") in new stack
-- Called g1/0452281211859
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Zap/1-1 answered Local/10452281211859@default-d007,2
> Channel Local/10452281211859@default-d007,1 was answered.
-- Executing Playback("Local/10452281211859@default-d007,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 10452281211859, 2) exited non-zero on 'Local/1045 2281211859@default-d007,2'
-- Executing DeadAGI("Local/10452281211859@default-d007,2", "agi://127.0.0.1 :4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----0") in new stack
-- Executing AGI("Zap/1-1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6-----ANSWER-----4-----0 completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("Zap/1-1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in ne w stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing AGI("Zap/1-1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Zap/1-1", "Zap/g1/92*168*001*125*8320||To") in new stack
Oct 19 01:02:09 NOTICE[8958]: app_dial.c:1076 dial_exec_full: Unable to create c hannel of type 'Zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (default, 192*168*001*125*8320, 3) exited non-zero on 'Zap/ 1-1'
-- Executing DeadAGI("Zap/1-1", "agi://127.0.0.1:4577/call_log--HVcauses--PR I-----NODEBUG-----34-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----3 4-----CONGESTION---------- completed, returning 0
-- Hungup 'Zap/1-1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/10455532329890@default-1f30,2", "agi://127.0.0.1:457 7/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/10455532329890@default-1f30,2", "Zap/g1/04555323298 90||To") in new stack
Oct 19 01:02:10 NOTICE[8972]: app_dial.c:1076 dial_exec_full: Unable to create c hannel of type 'Zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/10455532329890@default-1f30,2", "") in new stack
== Spawn extension (default, 10455532329890, 3) exited non-zero on 'Local/1045 5532329890@default-1f30,2'
-- Executing DeadAGI("Local/10455532329890@default-1f30,2", "agi://127.0.0.1 :4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in n ew stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----3 4-----CONGESTION---------- completed, returning 0


I know my zap channel is properly configured since I can dial out from asterisk, from a sip phone and manually through vicidial. I only changed my extensions.conf to reflect my local dial plan and to make the changes as per tutorial F on the managers manual.
I get the following as my list status:
STATUS STATUS NAME CALLED NOT CALLED
DROP Agent Not Available 4 0
NA No Answer AutoDial 1 0
SUBTOTALS 5 0
TOTAL 5

My box is a cheap atom pc with 1gb ram and an openvox a400p with only one FXO module.

Any help would be greatly appretiated since I am approaching my deadline.

[chrislog]

Feel free to ask me anything else you might need, I am going to be trying to figure this out all day, you'll hear back from me right away.

[mflorell]

We do not recommend using POTS FXO lines with ViciDial or Asterisk due to the horrible progress detection within Asterisk.

Have you tried this with SIP trunks?

[williamconley]

it looks like you made a call, and it worked, and you made another and the first one had not ended. so ... since you have a single trunk, you have no room for that second call.

does it work if you "pause" for a few second after each call to allow it to terminate properly?

[chrislog]

Hi, thank you for replying.
I was also comming to the conclusion that somehow the FXO was not signaling properly and that is why the line was getting congested. Is there A way I can make it "pause" as you said so that I can make it work? I have unlimited calls over POTSand I wanted to make use of that since I will be making a lot of calls. Otherwise, can I get this done with just one voip line? All my calls will be local (in mexico,tho). Know any provider that wouldnt, mind me calling twice a minute for 12 hours straight?

Also, I know that polititians and governments and etc, run this kind of campaigns all the time, do any of you know what they use?

Oh, also I forgot to mention that I was not able to make a call even if I had just one lead in the list, what gives?

[williamconley]

let me be sure we are on the same channel here.

I have unlimited calls

does not mean that you can make them all at once. if you have a single fxo port, you can only make one call at a time. you must wait for that call to terminate before you make another call.

before you attempt this in vicidial, you need to verify that it is working properly in asterisk.

using a local soft phone, dial out through the trunk and then hang up and be sure that it clears out properly.

before each step, run "show channels" at the asterisk CLI.

please also post your entire setup and avoid the word "Latest", as I do not want someone in april next year to guess what the version was this week. Please post your vicidial version and build, your asterisk version, zaptel version, how you installed (you said vicibox) and you've already listed your hardware. (a cheap atom pc with 1gb ram and an openvox a400p with only one FXO module).

you also need to read the manual (available on eflo.net for free, and you should seriously consider the paid version as soon as you get this running, it will save you quite a bit of time) and read the stickies for each forum before you post in it. (Those are the message threads listed at the beginning of each forum even though they are old, they are required reading before posting as is the manual.)