Solved: Vicidial keeps calling number non-stop

[mav2287]

ViciBox5.x86_64-5.0.3.preload from .iso | Vicidial: 2.8-380c BUILD: 130925-2119 | Asterisk 1.8.23 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel E6750 | Core Due 2.66

I am having issues with Vicidial detecting if a call is answered or ringing. When I have it call my cell phone to test it while the test cell phone is ringing it says Waiting for ring... and just keeps counts during the duration of the call. When I answer the phone it doesn't see that it was answered and still says waiting for ring.... and if I hang up the test cell phone vicidial calls back but doesn't show in the CLI. I can't even hang up through vicidial because it doesn't think there is a call on the line. I have read the manual and checked all over and I am still stumped any ideas?

Carrier is Vitelity and it is IP based registration so there is not user and pass

Account Entry:
[vitel-outbound]
type=friend
dtmfmode=auto
host=64.2.142.93
allow=all
canreinvite=no

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound32.vitelity,net ; I don't have a comma there but the forum thought I was spamming with the off site url
context=inbound
allow=all
insecure=port,invite
canreinvite=no

Dial Plan:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

cli asterisk -r


[Oct 6 14:07:341 == Using SIP RTP CoS .. ark 5
IOct 6 14:07:351 > Channel SIP/1234-00000006 was answered.
IOct 6 14 :07 :351 -- Executing 186000511Mefault: 11 MeetMe("SIP/1234-00000006", "8600051,F") in n
w stack
IOct 6 14:07:351 == Parsing' /etc/asterisk/ .. eetroe.conf': IOct 6 14:07:351 == Found
IOct 614:07:351 == Parsing '/etc/asterisk/ .. eetroe-uicidial.conf': IOct 614:07:351 == Found
IOct 6 14:07:351 -- Created MeetMe conference 1023 for conference '8600051'
IOct 6 14:07:351 -- <SIP/1234-00000006> Playing 'conf-onlyperson.gs .. ' (language 'en')
IOct 614:07:361 == Manager 'sendcron' logged off fro .. 127.0.0.1
IOct 614:07:381 == Manager 'sendcron' logged on fro .. 127.0.0.1
IOct 6 14: 07 : 38 I -- Execut i ng 186000511Mefau It: 11 MeetMe ("Loca 1/86000511Mefau I t-00000007 ; 2", "8
00051,F") in new stack
IOct 6 14:07:381 > Channel Local/8600051@default-00000007;1 was answered.
IOct 6 14: 07 : 38 I -- Execut i ng 191913593XXXX@default:l1 AG I ("Loca 1/8600051@default-00000007;1" ,
"agi://127.0.0.1:4577/call_Iog") in new stack
IOct 6 14:07:381 -- AGI Script Executing Application: (EXEC) Options: (SeH_CAMPCUST=MAltI))
IOct 6 14:07:381 -- <Local/8600051@default-00000007;1>AGI Script agi://127.0.0.1:4577/call_Iog
o"pleted, returning 0
IOct 614:07:381 -- Executing 191913593XXXX@default:21 Dial("LocaI/8600051@default-00000007;1",
"SIP/913593XXXX@uitel-outbound") in new stack
IOct 6 14:07:381 == Using SIP RTP CoS .. ark 5
IOct 6 14:07:381 -- Called SIP/913593XXXX@uitel-outbound
IOct 614:07:381 == Manager 'sendcron' logged on fro .. 127.0.0.1
IOct 6 14: 07 : 38 I -- Execut i ng 158600051@default:ll MeetMe ("Loca 1/58600051@default-00000008;2" ,
"8600051,F .. q") in new stack
IOct 6 14:07:381 > Channel Local/58600051@default-00000008;1 was answered.
IOct 6 14: 07 : 38 I -- Execut i ng 18309@default:ll Answer ("Loca 1/58600051@default-00000008;1", "")
in new stack
IOct 6 14: 07 : 38 I -- Execut i ng 18309@default:21 Mon i tor ("Loca 1/58600051@default-00000008;1", "wa
,20131006-130737_913593XXXX") in new stack
IOct 6 14: 07 : 38 I -- Execut i ng 18309@default:31 Wa i t ("Loca 1/58600051@default-00000008;1", "3600"
) in new stack
IOct 614:07:391 == Manager 'sendcron' logged off fro .. 127.0.0.1
IOct 614:07:391 == Manager 'sendcron' logged off fro .. 127.0.0.1
IOct 6 14:07:401 -- SIP/uitel-outbound-00000007 is .. aking progress passing it to Local/8600051@
efault-00000007;1
icidial><CLI>

[williamconley]

Oct 6 14: 07 : 38 I -- Execut i ng 18309@default:ll Answer

8309?

Perhaps you should check the "Routing Exten" in the campaign (press the ? next to the field).

[mav2287]

I went into the campaign and read the "?". I am assuming that the number you pointed out has to do with the "Campaign Rec exten:" as that is what was set to 8309. My routing exten is set to the default 8368. I thought that showed up because I had record set to "force all" and it was routing it to be recorded. I did try turning all recording options off and it still doesn't see that the line is picked up or ringing. The weirdest part is that it does dial the line rings and if someone answers you can talk to them but the agent screen will show "waiting for ring" and keep counting the whole time.

I also went in and created a new campaign using all the defaults and it does the same things. It is like it doesn't see the line ring and also doesn't see when someone picks up. I have double check my carrier setting and dial plan and tried various different variations. I have spent the day digging through the forum and reading but just can't figure out why the calls would go through buy vicidial wouldn't see it ring or see the person answer.

[williamconley]

I think i'm a little confused by your description of the events. I had thought (based on the ClI output) that you may have had 8309 for the routing extension by accident. Now I see it's being recorded ... so:

When you manually dial the phone number, vicidial initiates the recording, but does it ever actually transmit sound between your Lead and your Agent? Do they have a conversation?

The last entry posted is

Code: Select all

 IOct 6 14:07:401 -- SIP/uitel-outbound-00000007 is .. aking progress passing it to Local/8600051@
efault-00000007;1

Which means the call is not yet answered (that's what making progress means, it's been dialed but no response yet). So ... it's being monitored but hasn't been answered and likely other stuff happens after this. But you don't show that other stuff as this is the last line. We're missing the rest (like ... the call being answered), so it's hard to form an opinion of what happens then or what may go wrong.

[mav2287]

The interesting part is that I included everything to the point where I logout of the agent screen. I included another CLI and I went from login of agent, call by agent to logout of agent. Sorry if I did a poor job describing it I have been fighting with it most of the day and it is just plain weird. I really think it might be with the dial plan or carrier setup. I even tried a complete re-install, testing the setting on go-auto dial and a couple other things to try to rule out possibility. I get the same result will all of them so I think it must be carrier or carrier config related. I am using Vitelity which I know people use quite often so I don't know why it is giving me so much grief.

So to walk you through what it does. Lets say an agent logs in and manual dials someone the agent will hear the phone ring, the person will answer and conversation will start. The agent screen will still show "waiting for ring....." and it will be counting time. It will never see the call as answered and should the customer hang up almost instantly their phone will start ringing again and it will reconnect the call. I'm not sure what is happening on the technical side but it is like the system is waiting on the call to take place and doesn't realize the call is taking place. For example on the manual call I can hear it ringing but the system can't I know someone answered but it never shows that someone answered in the system.

If I have Vicidial set to autodial then it will dial the number but when the customer hangs up they just get dead air for about 11 seconds and then their line will go dead and their phone will ring again. If they answered again there is no one there and they do not get routed to an agent. That is on the same lead. So what you end up doing is spamming the phone with calls and it never connects to an agent.


Rasterisk CLI


[Oct 623:26:521 -- <Local/55558600051@default-00000003;2>AGI Script agi://127.0.0.1:4577/call
log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct 623:26:531 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:26:531 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:26:551 == Manager 'sendcron' logged on from 127.0.0.1
[Oct 623:26:551 == Using SIP RTP CoS mark 5
[Oct 623:26:571 > Channel SIP/201-00000007 was answered.
[Oct 6 23 :26 :571 -- Executing [8600051@default:ll MeetMe("SIP/201-0000000?", "8600051,F") in n
stack
[Oct 623:26:571 == Parsing' /etc/asterisk/meebe.conf': [Oct 623:26:571 == Found
[Oct 623:26:571 == Parsing '/etc/asterisk/meebe-uicidial.conf': [Oct 623:26:571 == Found
[Oct 623:26:571 -- Created MeetMe conference 1023 for conference '8600051'
[Oct 623:26:571 -- <SIP/201-00000007> Playing 'conf-onlyperson.gsm' (language 'en')
[Oct 623:26:591 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:27:021 == Manager 'sendcron' logged on from 127.0.0.1
[Oct 623:27:021 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:27:031 == Manager 'sendcron' logged on from 127.0.0.1
[Oct 623:27:041 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:27:071 == Manager 'sendcron' logged on from 127.0.0.1
[Oct 623:27:071 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:27:071 == Manager 'sendcron' logged on from 127.0.0.1
[Oct 6 23: 27 : 071 -- Execut i ng [8600051@default:ll MeetMe ("Loca ]/8600051@default-00000004;2", "
00051,F") in new stack
[Oct 6 23:27:071 > Channel Local/8600051@default-00000004;1 was answered.
[Oct 6 23: 27 : 071 -- Execut i ng [91913593XXXX@default:11 AG I ("Loca ]/8600051@default-00000004;1" ,
"agi://127.0.0.1:4577/call_Iog") in new stack
[Oct 6 23:27:071 -- AGI Script Executing Application: (EXEC) Options: (SeH_CAMPCUST=1001l)
[Oct 623:27:071 -- <Local/8600051@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_Iog
ompleted, returning 0
[Oct 623:27:071 -- Executing [91913593XXXX@default:21 Dial("Loca]/8600051@default-00000004;1",
"SIP/913593XXXX@uitel-outbound"To") in new stack
[Oct 6 23:27:071 == Using SIP RTP CoS mark 5
[Oct 623:27:071 -- Called SIP/913593XXXX@uitel-outbound
[Oct 623:27:081 == Manager 'sendcron' logged off from 127.0.0.1
[Oct 623:27:101 -- SIP/uitel-outbound-00000008 is making progress passing it to Local/8600051
efault-00000004;1
[Oct 623:28:031 == Manager sendcron logged on fro .. 127.0.0.1 [Oct 623:28:031 == Manager 'sendcron' logged off fro .. 127.0.0.1
[Oct 623:28:031 == Manager 'sendcron' logged on fro .. 127.0.0.1 [Oct 623:28:041 == Manager 'sendcron' logged off fro .. 127.0.0.1 [Oct 623:28:081 == Manager 'sendcron' logged on fro .. 127.0.0.1
[Oct 623:28:081 == Manager 'sendcron' logged off fro .. 127.0.0.1
[Oct 623:28:101 -- SlP/uitel-outbound-00000008 is .. aking progress passing it to Local/8600051@
efault-00000004;1
[Oct 623:28:271 == Spawn extension (default, 8600051, 1) exited non-zero on 'SlP/201-00000007' [Oct 623:28:271 -- Executing [h@default:11 AGl("SlP/201-0000000?", "agi://127.0.0.1:4577/call log--HVcauses--PRl-----MODEBUG-----16---------------") in new stack
[Oct 6 23:28:271 -- <SlP/201-00000007>AGl Script agi://127.0.0.1:4577/call_log--HVcauses--PRl--
--MODEBUG-----16--------------- co"pleted, returning 0
[Oct 6 23:28:371 -- SlP/uitel-outbound-00000008 answered Local/8600051@default-00000004;1
[Oct 6 23 :28 :501 -- Executing [h@default:11 AGl ("Loca]/8600051@default-00000004;1", "agi ://127 .
. . 0.1:4577/call_log--HVcauses--PRl-----MODEBUG-----16-----AMSWER-----103-----13") in new stack
[Oct 6 23:28:501 -- <Local/8600051@default-00000004;1>AGl Script agi://127.0.0.1:4577/call_log-
HVcauses--PRl-----MODEBUG-----16-----AMSWER-----103-----13 co"pleted, returning 0
[Oct 6 23:28:501 == Spawn extension (default, 91913593XXXX, 2) exited non-zero on 'Loca]/8600051@ efault-00000004;1'
[Oct 623:28:501 -- Hungup 'DAHDl/pseudo-1403239039'
[Oct 623:28:501 == Spawn extension (default, 8600051, 1) exited non-zero on 'Loca]/8600051@defau It-00000004;2'
[Oct 6 23 :28 :501 -- Executing [h@default:11 AGl ("Loca]/8600051@default-00000004;2", "agi ://127 .
. . 0.1:4577/call_log--HVcauses--PRl-----MODEBUG-----0---------------") in new stack
[Oct 6 23:28:501 -- <Local/8600051@default-00000004;2>AGl Script agi://127.0.0.1:4577/call_log-
HVcauses--PRl-----MODEBUG-----O--------------- co"pleted, returning 0
[Oct 6 23:29:031 == Manager 'sendcron' logged on fro .. 127.0.0.1
[Oct 623:29:031 == Manager 'sendcron' logged off fro .. 127.0.0.1
[Oct 623:29:031 == Manager 'sendcron' logged on fro .. 127.0.0.1
[Oct 623:29:031 == Manager 'sendcron' logged off fro .. 127.0.0.1
[Oct 623:29:081 == Manager 'sendcron' logged on fro .. 127.0.0.1
[Oct 623:29:081 == Manager 'sendcron' logged off fro .. 127.0.0.1

[mav2287]

SOLVED! Since I have read MANY a thread where the person looks like they solved it but didn't post the answer I thought I would post what I learned. Turns out it was not only an issue with my dial plan but with the config for the carrier. Below is what worked, I used IP registration so no username or passwords. William thanks for getting me headed the right direction.

Vitelity Outbound example

No registration string

[vitel-outbound]
type=peer
dtmfmode=auto
trustrpid=yes
sendrpid=yes
host=64.2.142.93
disallow=all
allow=g729
allow=ulaw
canreinvite=no

exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXNXXXXXX,2,Dial(SIP/vitel-outbound/${EXTEN:1},,tTor)
exten => _81NXXNXXXXXX,3,Hangup

Also check your Asterisk version in your Server admin page if these do not match it will not work. Apparently the way that Asterisk processes calls changed at some point and if Vicidial doesn't know the version it can't react accordingly. For example when mine was off it wouldn't see that a call was answered and would time out or just keep calling the number.

[dailafing]

I read this to the letter since I'm having the exact same issue, however with the addition of not hearing a ringing when dialling the prospect from the green manual dial screen.

Even though I read all if this, including the SOLVED post above, I'm still unsure what you did to fix this?
Are you saying you removed the registration string?

I'm at a loss at the moment, and my job could be on the line if I don't get thus issue fixed.
So please if anyone can spell out for me what is wrong and how to fix, I'd really appreciate it.

Thanks

[mav2287]

The first step will be to post your setup like I have in my signature. You also want to include info on what version of vicidial you have and how you installed it.

The problem for me was the extensions were wrong. I use ip authentication with my carriers so I could ditch the registration. You may also want to go in and check the asterisk version that shows in the server admin page. If it does not match what you have installed on your server it can also cause issues.

[williamconley]

I read this to the letter since I'm having the exact same issue, however with the addition of not hearing a ringing when dialling the prospect from the green manual dial screen.

Even though I read all if this, including the SOLVED post above, I'm still unsure what you did to fix this?
Are you saying you removed the registration string?

I'm at a loss at the moment, and my job could be on the line if I don't get thus issue fixed.
So please if anyone can spell out for me what is wrong and how to fix, I'd really appreciate it.

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 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. 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) Many people post "hey! I have the same problem!" but in the end it turns out that the CAUSE of "this doesn't work" is completely different in their situation than the original post. So ALWAYS post your specs even if you are certain you have The Same Problem.

4) Always actually describe your problem in your own words (to be sure we understand what YOUR problem is, especially if the post you are popping into has more than one entry, as the problem may "evolve" during multiple posts and we won't know which part applied to you!).

5) Without your Carrier configuration and some CLI output from your server, all we can really do is tell you to try the same fix the other guy tried. Which you've obviously tried (or you should turn in your keyboard now and get a nice job in the hospitality industry ... ).

Bring back some details, and we can walk through through a solution.

FYI: We ordinarily only charge $25 to configure a carrier. But we've found that many clients who believe that they have a functional server in need of a carrier have not even finished their installation yet! LOL (Luckily that does not happen nearly as much now that Vicibox and Goautodial are so popular!)

[dailafing]

Hi there, Lets see if I can give the information you need my friends!

Ok, looking at the boot up, i can see that the cli says....

- ViciBox Redux v.5.0.2-130821
myvicidial:~ # asterisk -r
Asterisk 1.8.23.1-vici, Copyright (C) 1999 - 2012 Digium, Inc. and others.

I installed using the iso from vicibox, it was the 32bit preloaded version...

I hope this complies with the requirements....

----------------------------------------------------------------------------------

That aside, I've worked out how to use putty, and now i know how people are able to copy and paste to the fourum!

This is the same issue that i had on another computer, and have since reinstalled on to another computer, using same disk, but as a troubleshooting tech, i didnt bother with the updates...

same issue...

-------------------------------------

so here is the CLI during the attempted phone call.....

Code: Select all

[Nov  7 09:10:04]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000004;2", "8600051,F") in new stack
[Nov  7 09:10:04]        > Channel Local/8600051@default-00000004;1 was answered.
[Nov  7 09:10:04]     -- Executing [075089MyMobileNumber@default:1] AGI("Local/8600051@default-00000004;1", "agi://127.0.0.1:4577/call_log") in new stack
[Nov  7 09:10:04]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=101))
[Nov  7 09:10:04]     -- <Local/8600051@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov  7 09:10:04]     -- Executing [075089MyMobileNumber@default:2] Dial("Local/8600051@default-00000004;1", "SIP/orbtalk_wmsx/075089MyMobileNumber,,tTor") in new stack
[Nov  7 09:10:04]   == Using SIP RTP CoS mark 5
[Nov  7 09:10:04]     -- Called SIP/orbtalk_wmsx/075089MyMobileNumber
[Nov  7 09:10:05] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #93)
[Nov  7 09:10:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:10:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:10:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:10:08]     -- SIP/orbtalk_wmsx-00000004 is making progress passing it to Local/8600051@default-00000004;1
[Nov  7 09:10:12]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:10:25] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #94)
[Nov  7 09:10:45] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #95)
[Nov  7 09:11:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:11:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:11:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:11:03]     -- SIP/orbtalk_wmsx-00000004 is making progress passing it to Local/8600051@default-00000004;1
[Nov  7 09:11:05] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #96)
[Nov  7 09:11:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:11:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:11:13]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:11:25] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #97)
[Nov  7 09:11:45] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #98)
[Nov  7 09:11:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:11:53]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000004;2'
[Nov  7 09:11:53]     -- Executing [h@default:1] AGI("Local/8600051@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov  7 09:11:53]     -- <Local/8600051@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov  7 09:11:53]   == Spawn extension (default, 075089MyMobileNumber, 2) exited non-zero on 'Local/8600051@default-00000004;1'
[Nov  7 09:11:53]     -- Executing [h@default:1] AGI("Local/8600051@default-00000004;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[Nov  7 09:11:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:11:53]     -- <Local/8600051@default-00000004;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
[Nov  7 09:11:54]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:11:54]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:12:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:12:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:12:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:12:05] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #99)
[Nov  7 09:12:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Nov  7 09:12:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:12:12]   == Manager 'sendcron' logged off from 127.0.0.1
[Nov  7 09:12:25] NOTICE[2169]: chan_sip.c:13795 sip_reg_timeout:    -- Registration for 'MySipUserName@192.104.103.6' timed out, trying again (Attempt #100)


On this occasion, i let my mobile ring through to answer-phone, and the dialler retried the number within seconds, because i hadn't hit the hand up button? would have expected it to give up?

So the issues are:
1) It's not producing a ringing tone while dialling the number, waiting for the answer, or answerphone.
2) When user answers, it does not recognise it, and it still says 'Waiting for ring', and counts upwards...
3) when i then hang up on the number, the 'tone' is played (std gsm sound) but the system is still saying 'waiting for ring'...

after a few minutes, i receive an error on the agent side.... looks like...


So im hoping that this all helps someone help me out...

I read the easily googlable 'vicidial for dummies', but quickly out grew that once i realised that its not related to the iso that i used...
So then I followed the managers manual for my installation....

I hope you can help me out, thanks

[mav2287]

What are the carrier settings?

[dailafing]

Knew I'd Forgotten Something... I'm Trying My Best

[mav2287]

I would knock out that registration timeout first does your carrier offer IP authentication vs. registration? If they do switch to it and see if that solves the problem.

Also does your provider allow dialers? I tried to get ringcentral setup as a carrier and no matter what I tried it wouldn't work.

[dailafing]

I would knock out that registration timeout first does your carrier offer IP authentication vs. registration? If they do switch to it and see if that solves the problem.

Also does your provider allow dialers? I tried to get ringcentral setup as a carrier and no matter what I tried it wouldn't work.

Hi there, thanks for your help.

Yes, I've spoken in depth with the carrier and they have in fact sold me a package specifically for call centres.

Where can I find the registration timeout? Thanks

[stahldrache]

Please check your asterisk version of server in admin page, it should be 1.8.23.1-vici. If not, update it and reboot.

[dailafing]

Yes I'm running 1.8.23.1-vici

[mav2287]

Have you tried to setup a soft phone with your carrier information to see if it at least works? It is an easy way to test a carrier

[dailafing]

Yes, I used 3cx, and you can clearly hear the ringing. It's getting lost somewhere in the dialler.

I'm expecting this to have been a common issue that every old hand would know how to fix?

I really need to get this going soon since my boss was expecting a test dial this week.

Thanks

[mav2287]

If you are working on a deadline and really need this done it may be worth your time and money to bring in a pro to knock this out. From everything I have read and my dealings with William I personally would call the Pound Team (http://www.poundteam.com). I believe they charge $25 to setup a carrier if I remember correctly from a previous post, but I don't know for sure. I can also tell you from experience with William the he really knows his stuff when it comes to vicidial.

Poundteam Incorporated
306 Fox Loop
Davenport, FL 33837
(352) 269-0000
Fax: 920-260-0000
Toll Free: 888-883-8488

THe problem is that you can chase a problem for days get nowhere then have to wait more days for a reply on the forum then you find you have 3 more issues you didn't even realize you had. To be honest with you at this point your problem appears so different form mine I really don't know what else to tell you to look for that would be causing this.

[dailafing]

I know what you mean. But in hoping to get an understanding of this myself so I can fix things like this as they pop up.
The fact is, we've only just started up and we need to show a return using this hosted dialler before anything extra can be spent.

[williamconley]

Have you tried to setup a soft phone with your carrier information to see if it at least works? It is an easy way to test a carrier

If the server is already dialing successfully through the carrier, that's only necessary as a "down server" scenario (ie: server is down, you can still dial through the carrier and even receive calls, kinda handy).

I would knock out that registration timeout first does your carrier offer IP authentication vs. registration? If they do switch to it and see if that solves the problem.

Also does your provider allow dialers? I tried to get ringcentral setup as a carrier and no matter what I tried it wouldn't work.

Not related to registration at all. interesting concept, though. Registration failing generally kills INBOUND calls but has no relation to outbound (as a rule).

If your server continues to dial a number it is because it is recycling the lead in some fashion or does not believe it actually dialed it (so it dials it ... to you it's "again" but to it it's just dialing).

Turn OFF lead recycling. Also, don't use it for at least 12 months, preferably never. There is not presently a viable scenario where lead recycling is useful.

If that does not solve your problem, you'll need to find out why the dialer believes it should be dialing the lead again. Check your "available leads" and your hopper to see if it actually Loads that lead again. Best bet would be to find one that it happens on, create a "fake" carrier (that only dials a local SIP phone, so as not to annoy the client in question), and then make that single lead the only lead in the campaign and see if you can figure out what is really happening.

A couple extra points to ponder: _X. is a BAD extension. Let's put a real dialplan entry in that in some fashion describes the dialplan rules for your country. USA = NXXNXXXXXX if you add the dial prefix from the campaign "9" and the dial code "1", you end up with _91NXXNXXXXXX and you change the dial command to include an EXTEN:1 instead of EXTEN so it drops the 9 (thus using the 9 to choose the carrier, allowing you to have multiple carriers per server all active simultaneously). In the US the dial code would generally be "0" instead of "1", and I believe the 4th digit CAN be a 0 or 1, so _90NXXXXXXXXX would be likely. If you look in extensions.conf you'll also see some dialplan entries designed to capture "bad" calls (wrong number of digits and such) to kill them immediately and generate a new call. But _X. is just a bad idea, it can capture calls it should not be able to and break functionality in Vicidial.

There are plenty of sample dialplan entries around.

[Joss2103]

Hello, someone has found a solution, I have the same problem, the calls keep remarking

help me please!!