vicidial.org

Once in a while the following alert is displayed to the agent one Manual Dial:
"Dial timed out, contact your system administrator"

The problem is intermittent, it does not seem to be related to the dialplan or the number called, because it does not occur when calls to the same numbers are repeated.

I've observed the problem in calls through PRI and SIP.

Has anyone else in the list detected a similar issue? In what cases might this error message be reported?

Vicidial 2.2 rev. 1240
Asterisk 1.2.30.2

Upgrade to SVN from today

The problem should go away with all of the new manual dial code I just added.

Thanks Matt, this is great news. I checked the Bug Tracker today, and I've seen that you've been hammering out bugs at an incredible pace.

I have spent quite a bit of time on those bugs in the last two days, I haven't touched the volume buttons or gettext issues though, and I'm not sure I can solve the volume issue right now either, but that is relatively minor.

Hi Matt,

It is a long weekend in Colombia. Monday is holiday. I will be working all day on updating the gettext patch for the last revision of 2.2 in SVN.

The volume buttons bug is a minor issue, but I feel sooner or later that a definitive solution for tracking outbound call channels on Manual Dial will be needed. As far as I understand the following issues have to do with channel tracking on Manual Calls:
http://www.eflo.net/VICIDIALmantis/view.php?id=248
http://www.eflo.net/VICIDIALmantis/view.php?id=249
http://www.eflo.net/VICIDIALmantis/view.php?id=274
http://www.eflo.net/VICIDIALmantis/view.php?id=271

PS: I know you've just posted a work-arounds for issues 248, 274 in SVN. I will do some testing this week to inform you of the results.

Thanks

hi matt, i have installed 2.2 on svn for while on its early developement stage, and im suing it for production.how do i update with your latest 2.2 SVN..?

mkdir /usr/src/astguiclient
cd /usr/src/astguiclient
svn checkout svn://svn.eflo.net:43690/agc_2-X/branches/agc_2.2.0

do i need to re-compile the asterisk again...before setting up the new SVN 2.2.

Sheesh Ruben23! No! Have you found any documentation regarding recompiling Asterisk when upgrading Vicidial? You already have more than 300 posts and you still haven't grasp the basics of Vicidial.

Asterisk recompile is done when people are waiting to work and there is a deadline.

And the compiler is given the lowest priority on the server.

Hi,

What's the solution for the Dial Timed out error?

Thanks,

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... 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 "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

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 | Intel DG35EC | Core2Quad Q6600

3) Dial timeout is not an error: It's a notification of an event. You called, no one answered.

4) Now, the REASON no on answered may show a problem with the system and needs resolution. For that: Please post an asterisk CLI output of a single example (not 3000 lines of unrelated code, just a single call with ONLY that traffic on the dialer when the test is run).

Happy Hunting

3) Dial timeout is not an error: It's a notification of an event. You called, no one answered.

4) Now, the REASON no on answered may show a problem with the system and needs resolution. For that: Please post an asterisk CLI output of a single example (not 3000 lines of unrelated code, just a single call with ONLY that traffic on the dialer when the test is run).

Happy Hunting

Hi Marcelo here

- ViciBox v.9.0.0 190913-1108 * Released on Friday the 13th during a full moon. So spooky, much wow! |Vicidial 2.14-588c BUILD 190925-1346 | Asterisk 13.27.0-vici | Linux version 4.12.14-lp151.28.16-default | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Xeon(R) CPU E5-2450 0 @ 2.10GHz

> > > > > > > > > > I am having the same answer from Vicibox when running my first test campaign.

> > > > > > > > > > Calls posted direct from my xLite goes out normally:
Dialplan entry for carrier Langineers:

:exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(SIP/es1.langineers.com:5060/${EXTEN:1})
exten => _1NXXNXXXXXX,3,Hangup

exten => _XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXX,2,Dial(SIP/es1.langineers.com:5060/${EXTEN})
exten => _XXXXXXX,3,Hangup

exten => _NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXNXXXXXX,2,Dial(SIP/es1.langineers.com:5060/${EXTEN})
exten => _NXXNXXXXXX,3,Hangup


> > > > > > > Call log when dialing direct from the xLite for phone number 4157066724 [NOTE: also works for calls to 14157066724 and 1234567 no problem]
[Oct 11 16:08:02] WARNING[580]: chan_sip.c:4128 retrans_pkt: Timeout on 1103300734-2069980851-1726045714 on non-critical invite transaction.
[Oct 11 16:08:03] == Using SIP RTP CoS mark 5
[Oct 11 16:08:03] > 0x7fd7901b4650 -- Strict RTP learning after remote address set to: 192.168.15.194:56638
[Oct 11 16:08:03] -- Executing [4157066724@default:1] AGI("SIP/201-0000007a", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 11 16:08:03] -- <SIP/201-0000007a>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 11 16:08:03] -- Executing [4157066724@default:2] Dial("SIP/201-0000007a", "SIP/es1.langineers.com:5060/4157066724") in new stack
[Oct 11 16:08:03] == Using SIP RTP CoS mark 5
[Oct 11 16:08:03] -- Called SIP/es1.langineers.com:5060/4157066724
[Oct 11 16:08:05] > 0x7fd7b80138c0 -- Strict RTP learning after remote address set to: 64.124.219.133:17826
[Oct 11 16:08:05] -- SIP/es1.langineers.com:5060-0000007b is making progress passing it to SIP/201-0000007a
[Oct 11 16:08:05] > 0x7fd7b80138c0 -- Strict RTP switching to RTP target address 64.124.219.133:17826 as source
[Oct 11 16:08:06] == Spawn extension (default, 4157066724, 2) exited non-zero on 'SIP/201-0000007a'
[Oct 11 16:08:06] -- Executing [h@default:1] AGI("SIP/201-0000007a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------------SIP 183 Session Progress)") in new stack
[Oct 11 16:08:06] -- <SIP/201-0000007a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 183 Session Progress) completed, returning 0


> > > > > > > Fact: From vicibox
Logged in as User: marcelo_agent on Phone: SIP/201 to campaign: TESTCAMP
Running manual calls from the lead we have.
It does not show anything on Asterisk console <<<<<<<<<<<< But the message "Dial timed out, contact your system administrator" shows on Vicibox.

Can we check Vicibox logs?

> > > > > > Please advise.

what is the dial prefix, manual dial prefix and (just for fun) the 3-way dial prefix on that campaign? If you research those options you may find your problem. During manual dial I believe there is also a notice of what the dial prefix will be for confirmation.

Edit after publishing:

> > > > > > > NEVER MIND -- I have fixed by adding entry on Carrier ID: Langineers' Dial Plan

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/es1.langineers.com:5060/${EXTEN:2})
exten => _91NXXNXXXXXX,3,Hangup

It is a work around as the campaign prefix was edited to "blank" from "9"

Many thanks.

I am trying to understand the hopper better, and may have more questions later.

Regards,

Marcelo

williamconley wrote:what is the dial prefix, manual dial prefix and (just for fun) the 3-way dial prefix on that campaign? If you research those options you may find your problem. During manual dial I believe there is also a notice of what the dial prefix will be for confirmation.

I have checked and the "dial prefix" was "9" and I left blank.
Manual dial prefix and 3-way dial prefix was left blank.

I do not think the problem is here, as Asterisk does NOT show any call made from the campaign, nor good nor bad with "9" prefix, already fixed.

One other problem on top here < < < < < << < < < < < < < < < This is preventing me from testing.

My agent is locked out: "Sorry, there are no leads in the hopper for this campaign"

I have very few listed numbers and my campaign although still have 19 "NEW New Lead" using new and already called numbers, still "This campaign has 0 leads in the dial hopper."

I cannot try testing anymore, leads I have added to the list has duplicate numbers.

> > > > > > > > > > Question: How can I "force" a "NEW lead" be pushed into the dial hopper? < < < < < < < <
I my case I have 19 new leads of already called numbers and no leads in the hopper.

Hopper configuration:

Minimum Hopper Level: 5
Automatic Hopper Level: N
Automatic Hopper Multiplier: 1
Auto Trim Hopper: N
Hopper VLC Dup Check: N
Manual Dial Hopper Check: N
Force Reset of Hopper: N

Allow No-Hopper-Leads Logins: N
No Hopper Dialing: N


Please advise.

P.S. Edit after publishing:

When getting "Dial time out, contact your system administrator"

I am getting the following and NO Asterisk log.

"STATUS: Calling: (415)651-4087 UID: M0141150030000000008 Waiting for Ring... 1 seconds"


After another MANUAL attempt after cleaning the hopper:

STATUS: Calling: (415)706-6724 UID: M0141205330000000010 Waiting for Ring... 38 seconds

Vicibox log vdautodial.2019-10-14 shows:

2019-10-14 12:06:02|| dead call vac XFERd do nothing|10|4157066724|XFER||
2019-10-14 12:06:04|| dead call vac XFERd do nothing|10|4157066724|XFER||
2019-10-14 12:06:07|| dead call vac XFERd do nothing|10|4157066724|XFER||
2019-10-14 12:06:09|| dead call vac XFERd do nothing|10|4157066724|XFER||


action_send.2019-10-14 log file shows

2019-10-14 12:05:33|1|36|
Action: Originate
Exten: 914157066724 < < < < < < < < < < < This still show the "9" dial prefix.
Context: default
Channel: Local/8600051@default/n
Priority: 1
Callerid: "M0141205330000000010" <0000000000>
Timeout: 60000

Please Help me too

I have recently did a fresh install of ViciBox v.10.0.1 220503, here is my vicidial configuration on it's GUI attached below.

https://drive.google.com/file/d/12jjSOt ... sp=sharing

port used is 5060, callcentric is the SIP provider.

output of extension status is below

Code: Select all

asterisk -rx "sip show peer 1777HIDDEN" | grep Status

Code: Select all

Status : OK (15 ms)

kindly help !

zaurzo wrote:Please Help me too

I have recently did a fresh install of ViciBox v.10.0.1 220503, here is my vicidial configuration on it's GUI attached below.

https://drive.google.com/file/d/12jjSOt ... sp=sharing

port used is 5060, callcentric is the SIP provider.

output of extension status is below

Code: Select all

asterisk -rx "sip show peer 1777HIDDEN" | grep Status

Code: Select all

Status : OK (15 ms)

kindly help !

Please post the asterisk CLI output. Also contact your carrier to see if they are actually getting the call (and if they are denying it ... why?)