vicidial.org

Hi,

I am a newbie both for Linux and Vicidial.

My system setup is:
Vicidial Redux 64bit
Corei7, 16Gb RAM, 1TB HD
Softphones (Eyebeam), no hardphones

I have successfully installed the Redux version. I don't know if this information is significant, but my install only work when I boot under failsafe. Otherwise, it just shows a blank screen.. However, this is not the problem I need help on. I just included the info just in case it may be related....

Here is the thing, when do calls with just the softphones directly, without logging on to the Agent web client, the call lines are clear and does not have any problem. However, when the call is directed via the web client, the problem occurs. The line is jittery/choppy. You will immediately notice this once Vicidial connects to the softphone.. the message already goes like, "yoouuuu uuuu aaaa-aaarrrreeeee da-daaa-theeee ooooo-ooonnnlllyyyy..." This is the same during live calls.

Anyone has any idea on how to fix this? CLI does not show anything except that the call is connected, so there's not a lot of info to go with.
Any help is appreciated!

1) Welcome to the Party!

2) I note that you are trying to post your pertinent system info, this should help:

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

3) You should post (even if virtually identical) the asterisk CLI output from both a manual (nice sounding) and "directed via web client" (nasty sounding) single call. (Not 3000 lines of code, no other traffic, just one call from each sample).

4) Do you have auto-dial working on this system? Or are these "direct via the web client" calls Manual dial calls?

5) You may also want to post your carrier settings (mask user/pass/domain by replacing them with "user" ...)

6) In case we do get this going for you, it will be very helpful if you post your actual motherboard model number and any custom settings you've had to put in to make the motherboard work.

7) However, I will note that having to run on failsafe is not a good sign. I'd actually advise trying to install again. Perhaps trying the 32-bit just to be funny ... and even try GoAutoDial if you can to see if you can get a good clean installation with any available .iso. From there everything else seems to work much better, but any base install problems rarely work out well.

Please also verify (just to be funny) that you are installing this directly in the i7 and not in a virtual environment running on the i7.

Vicibox Redux 3.1 from .iso | Vicidial 2.4-310a Build 110506-1537| Asterisk 1.4.39.2-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

motherboard: Gigabyte H67MA, utilizing onboard audio/video.
Installation is NOT VM

Tried 32bit install previously same problem. The 64bit actually had a bit of improvement as the jitter seemed a bit shorter compared to the former.

Will be trying goautodial tomorrow but I still hope to get some idea on fixing this problem as I might (hopefully not) encounter the same thing during installation of a different iso.[/img]

Thanks for posting your specs.

FYI: unless I misunderstand something about your setup, you are NOT using "onboard audio/video". The onboard audio/video system is for use with local speakers and monitors. You are using neither. All audio processes are being passed through your system and managed by asterisk with appropriate software drivers.

I have not personally had a lot of luck with gigabyte boards, but it was ordinarily a stability issue rather than an odd sound issue.

I also note that you mentioned jitter and this makes me a bit nervous. ordinarily anyone who mentions a technical detail such as that during a vicidial installation is an asterisk technician who (inevitably) creates a problem by "knowing more than the installation manual and doing it HIS way instead."

To counter this feeling, please assure me that you have done everything in the Vicidial Manager's Manual exactly as written, without altering it for how "you know" asterisk works ... (obviously after it's up and running smoothly such changes are easily verified not to do harm ... but until you reach smoothness .... well, you get the idea, right?)

On that note, I see you did not post your carrier settings.

And now that I think of it, if you are experiencing sound issues while hearing "you are the only person in this conference" I'll need to ask about your network between your agent and the Vicidial system, and the protocol you are using to connect them.

Did I say welcome to the party already?

Hi Again,

Followed everything to the dot on the manual as i am fairly new to both linux, voip and vicidial, i have no other way of working it. By jitter, i guess that's the only description i can provide with the way i hear the the welcome voice from the conference and the person at the otherside of a test call.

i know 'asterisk' works as assumption that the call is pushing through if not logged on to the agent web client. The softphone is setup to connect to the asterisk part of the server, right?

[siptrunk]
disallow=all
allow=ulaw
type=friend
host=xx.xx.xx.xx
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TESTSIPTRUNK}/3272${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

looks fairly normal.

how about asterisk CLI from both manual and auto (not 3000 lines of code, just a single call from each scenario with no other mixed-in traffic)

OK.. I installed GoAutoDial today. Everything seems to be ok but I still got problems. I don't have audio. Calls ring twice and reflects as connected but there's no audio, I think for both ends. It is the same with and without being on the agent web client.

CLI capture below.

Carrier Setting is the same as above. Only difference is, I am now on

GoAutoDial CE 2.1 from .iso | Vicidial 2.4-309a Build 110430-1642 | Asterisk 1.4.39.1-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

[Sep 12 23:23:05] -- Executing [916047331918@default:1] AGI("SIP/8001-0000001d", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 12 23:23:05] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 12 23:23:05] -- Executing [916047331918@default:2] Dial("SIP/8001-0000001d", "SIP/tritel/327216047331918||tTor") in new stack
[Sep 12 23:23:05] -- Called tritel/327216047331918
[Sep 12 23:23:06] == Parsing '/etc/asterisk/manager.conf': [Sep 12 23:23:06] Found
[Sep 12 23:23:06] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 12 23:23:06] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 12 23:23:08] -- SIP/tritel-0000001e is making progress passing it to SIP/8001-0000001d
[Sep 12 23:23:09] -- SIP/tritel-0000001e answered SIP/8001-0000001d
[Sep 12 23:23:15] -- Executing [h@default:1] DeadAGI("SIP/8001-0000001d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----6") in new stack
[Sep 12 23:23:15] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----6 completed, returning 0
[Sep 12 23:23:15] == Spawn extension (default, 916047331918, 2) exited non-zero on 'SIP/8001-0000001d'



When I was with ViciBox, I didn't have the firewall setup but I forwarded all the necessary ports and port-ranges. With GoAutoDial, firewall is up and active, and I am confused of its settings. Do I have to change any of it?

first try turning it off to see if that is your issue.

if your server is behind a firewall... try to put the real IP address in sip.conf

open /etc/asterisk/sip.conf

externip = XX.XX.XX.XX --> IP of the firewall/router

it happens to me while using Tritel/Novatel carrier.

hope this helps...

how do you turnoff the firewall in goautodial?