vicidial.org

OK so maybe someone can figure this out, If I log into vicidial agent login with an IAX2 phone, i hear "your the only one...." and then "buzz", and then it basically seems the VD conference isnt opening /connecting the agent to a "channel" in order to make the call and If i log into it with a SIP phone, i dont hear anything. But If I log into the SIP phone and then dial the iax phone, i can hear "youve been kicked from this conference" when i logout. I also notice that it produces a different type of channel when i "show conferences in channel" it looks like the following:

LIVE CALLS IN YOUR SESSION:
# REMOTE CHANNEL HANGUP VOLUME
1 Local/8600052@default-00000037;2 HANGUP
2 SIP/1000-00000012 HANGUP

Default-000000037 which seems like the "channel" i would need all calls to connect to in order for them to actually ring....

Asterisk CLI:

Code: Select all

--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x20ace70 -- Strict RTP learning after remote address set to: my ip:8000
Peer audio RTP is at port my ip:8000
sip_route_dump: route/path hop: <sip:1000@myip:5060;rinstance=c822f72950dfd92d;transport=UDP>
Transmitting (NAT) to my ip:5060:
ACK sip:1000@my ip:5060;rinstance=c822f72950dfd92d;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK0e80919a;rport
Max-Forwards: 70
From: "S2004211344328600052" <sip:7542438008@172.26.15.181>;tag=as3a65704d
To: <sip:1000@my ip:5060;rinstance=c822f72950dfd92d;transport=UDP>;tag=3f304344
Contact: <sip:7542438008@172.26.15.181:5060>
Call-ID: 55429e315946d3743ba046db58518a41@172.26.15.181:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.21.0-vici
Content-Length: 0


---
    -- SIP/1000-00000011 answered
    -- Executing [8600052@default:1] MeetMe("SIP/1000-00000011", "8600052,F") in new stack
    -- Created MeetMe conference 1023 for conference '8600052'
    -- <SIP/1000-00000011> Playing 'conf-onlyperson.gsm' (language 'en')
  == Manager 'sendcron' logged off from 127.0.0.1
Reliably Transmitting (NAT) to my ip:33635:
OPTIONS sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK32009569;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as25558d1a
To: <sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP>
Contact: <sip:asterisk@172.26.15.181:5060>
Call-ID: 5c6f01e618b547907dc4f7a559cee1a9@172.26.15.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 21 Apr 2020 17:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to my ip:33635:
OPTIONS sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK32009569;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as25558d1a
To: <sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP>
Contact: <sip:asterisk@172.26.15.181:5060>
Call-ID: 5c6f01e618b547907dc4f7a559cee1a9@172.26.15.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 21 Apr 2020 17:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to 192.76.120.10:5060:
OPTIONS sip:sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK225fb0d2;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as6f2ee7e6
To: <sip:sip.telnyx.com>
Contact: <sip:asterisk@172.26.15.181:5060>
Call-ID: 534b63701ee2317c06f0fdc7040064f0@172.26.15.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 21 Apr 2020 17:44:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.76.120.10:5060 --->
SIP/2.0 200 Keepalive
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK225fb0d2;rport=5060;received=3.218.149.105
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as6f2ee7e6
To: <sip:sip.telnyx.com>;tag=dfb4940bfc7117e4d7fa62ed6ef36d37.3f98
Call-ID: 534b63701ee2317c06f0fdc7040064f0@172.26.15.181:5060
CSeq: 102 OPTIONS
Server: kamailio (5.0.8 (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '534b63701ee2317c06f0fdc7040064f0@172.26.15.181:5060' Method: OPTIONS
Retransmitting #2 (NAT) to 73.46.29.100:33635:
OPTIONS sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK32009569;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as25558d1a
To: <sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP>
Contact: <sip:asterisk@172.26.15.181:5060>
Call-ID: 5c6f01e618b547907dc4f7a559cee1a9@172.26.15.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 21 Apr 2020 17:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Called 8600052@default
    -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000035;2", "8600052,F") in new stack
    -- Local/8600052@default-00000035;1 answered
    -- Executing [300@default:1] Dial("Local/8600052@default-00000035;1", "IAX2/cc300,60,") in new stack
    -- Called IAX2/cc300
    -- Call accepted by 73.46.29.100:4570 (format ulaw)
    -- Format for call is (ulaw)
    -- IAX2/cc300-4491 is ringing
--
  == Manager 'sendcron' logged off from 127.0.0.1
Retransmitting #4 (NAT) to my ip:33635:
OPTIONS sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK32009569;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.26.15.181>;tag=as25558d1a
To: <sip:1110@10.0.0.55:33635;rinstance=54cdf0529f73f5a8;transport=UDP>
Contact: <sip:asterisk@172.26.15.181:5060>
Call-ID: 5c6f01e618b547907dc4f7a559cee1a9@172.26.15.181:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.21.0-vici
Date: Tue, 21 Apr 2020 17:44:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '5c6f01e618b547907dc4f7a559cee1a9@172.26.15.181:5060' Method: OPTIONS
    -- IAX2/cc300-4491 answered Local/8600052@default-00000035;1
    -- Channel IAX2/cc300-4491 joined 'simple_bridge' basic-bridge <452ff547-fe12-48ba-bd66-3b7463ad5634>
    -- Channel Local/8600052@default-00000035;1 joined 'simple_bridge' basic-bridge <452ff547-fe12-48ba-bd66-3b7463ad5634>

<--- SIP read from UDP:my ip:5060 --->


<------------->
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/1000-00000011
  == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/1000-00000011'
    -- Executing [h@default:1] AGI("SIP/1000-00000011", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[Apr 21 13:44:53] WARNING[17354][C-000000e2]: res_agi.c:2039 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)': Connection refused
[Apr 21 13:44:53] WARNING[17354][C-000000e2]: res_agi.c:2104 launch_netscript: Couldn't connect to any host.  FastAGI failed.
Scheduling destruction of SIP dialog '55429e315946d3743ba046db58518a41@172.26.15.181:5060' in 10112 ms (Method: INVITE)
Reliably Transmitting (NAT) to my ip:5060:
BYE sip:1000@my ip:5060;rinstance=c822f72950dfd92d;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.26.15.181:5060;branch=z9hG4bK51da2109;rport
Max-Forwards: 70
From: "S2004211344328600052" <sip:7542438008@172.26.15.181>;tag=as3a65704d
To: <sip:1000@my ip:5060;rinstance=c822f72950dfd92d;transport=UDP>;tag=3f304344
Call-ID: 55429e315946d3743ba046db58518a41@172.26.15.181:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.21.0-vici
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

    -- Called 55558600052@default
    -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-00000036;2", "8600052,K") in new stack
    -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-00000036;2", "") in new stack
  == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-00000036;2'
[Apr 21 13:44:53] WARNING[17393][C-000000e5]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/55558600052@default-00000036;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Apr 21 13:44:53] WARNING[17393][C-000000e5]: res_agi.c:2039 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection refused
[Apr 21 13:44:53] WARNING[17393][C-000000e5]: res_agi.c:2104 launch_netscript: Couldn't connect to any host.  FastAGI failed.
    -- <Local/8600052@default-00000035;2> Playing 'conf-kicked.gsm' (language 'en')

<--- SIP read from UDP:73.46.29.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP my ip:5060;branch=z9hG4bK51da2109;rport=5060;received=server ip
Contact: <sip:1000@7my ip:5060;rinstance=c822f72950dfd92d;transport=UDP>
To: <sip:1000@my ip:5060;rinstance=c822f72950dfd92d;transport=UDP>;tag=3f304344
From: "S2004211344328600052"<sip:7542438008@172.26.15.181>;tag=as3a65704d
Call-ID: 55429e315946d3743ba046db58518a41@172.26.15.181:5060
CSeq: 103 BYE
User-Agent: Zoiper rev.6751
Content-Length: 0

    -- Hungup 'DAHDI/pseudo-857186018'
    -- Executing [8600052@default:2] Hangup("Local/8600052@default-00000035;2", "") in new stack
  == Spawn extension (default, 8600052, 2) exited non-zero on 'Local/8600052@default-00000035;2'
[Apr 21 13:44:55] WARNING[17372][C-000000e3]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/8600052@default-00000035;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Apr 21 13:44:55] WARNING[17372][C-000000e3]: res_agi.c:2039 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection refused
[Apr 21 13:44:55] WARNING[17372][C-000000e3]: res_agi.c:2104 launch_netscript: Couldn't connect to any host.  FastAGI failed.
    -- Channel Local/8600052@default-00000035;1 left 'simple_bridge' basic-bridge <452ff547-fe12-48ba-bd66-3b7463ad5634>
  == Spawn extension (default, 300, 1) exited non-zero on 'Local/8600052@default-00000035;1'
[Apr 21 13:44:55] WARNING[17371][C-000000e4]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
    -- Executing [h@default:1] AGI("Local/8600052@default-00000035;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----11-----)") in new stack
[Apr 21 13:44:55] WARNING[17371][C-000000e4]: res_agi.c:2039 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----11-----)': Connection refused
[Apr 21 13:44:55] WARNING[17371][C-000000e4]: res_agi.c:2104 launch_netscript: Couldn't connect to any host.  FastAGI failed.
    -- Channel IAX2/cc300-4491 left 'simple_bridge' basic-bridge <452ff547-fe12-48ba-bd66-3b7463ad5634>
    -- Hungup 'IAX2/cc300-4491'
Really destroying SIP dialog '43688767772f71b127470af82a6a79c1@172.26.15.181:5060' Method: OPTIONS


Dialplan directly from Telnyx in regards to Vicidial:

exten => _9NXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXXXXXX,2,Dial(${Telnyx}/${EXTEN:1},60,tTor)
exten => _9NXXXXXXXXXX,3,Hangup

I feel like its prolly something stupid at this point like leaving "use 1" before dialing someplace or dialplan is wrong.

Heres the link to my dialplan: https://support.telnyx.com/en/articles/ ... l-ip-trunk


My question is, shouldnt I be able to use manual dial to call my cell, or override like i did to dial another extension? And if i had the carrier setup wrong, would i still be able to register my softphone thru it? Of which ive tried, xlite, zoiper free, zoiper 3 and zoiper 5, and bria....

Can someone PLEASE PLEASE PLEASE help me solve this because I already tried to just avoid the method I used which is a scratch install from the "docs" in vicibox9 on opensuse 15.1 via AWS Lightsail because I believe I can cluster on lightsail using their new database servers if i could just get the friggin dialer running, lol.(high hopes i know) But like I was saying I tried to avoid that altogether and install from iso using the new vicibox9.0.2 using VirtualBox and it doesnt fully install that way, for some reason the install doesnt finish, it hangs on a core2quad with 8gb ram and 200ssd.....even if i try to just do vicibox-express, it always dies at telephony server portion and then trying to manually do everything from there kinda defeats the purpose even though ive been trying all night.

1) Please try to use a subject line that actually provides a subject. Everybody on the support forum is here because they couldn't find the solution without the forum.
2) You did not post your installation information. Vicibox? Goautodial? A link to the installation instruction set you used? These are all very important if you are having a problem getting the system online. In fact, without that information it could take weeks to wander step-by-step through "how to get the system working" with back and forth. But knowing where we have started makes a huge difference. Huge.
3) Network topology is important here. Is the IAX2/SIP workstation on the same subnet as the Vicidial server? Are you accessing via public or private IP? Is NAT involved?
4) Why are we discussing a carrier during your login? Telephone company is not related to a login via SIP or IAX2.
5) Have you verified that your sip.conf's "externip" value is set to your public IP?
6) This is troublesome and makes me want to know the answer to question #2 even more:

Code: Select all

[Apr 21 13:44:53] WARNING[17393][C-000000e5]: res_agi.c:2039 handle_connection: Connecting to '127.0.0.1:4577' failed for url 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)': Connection refused
[Apr 21 13:44:53] WARNING[17393][C-000000e5]: res_agi.c:2104 launch_netscript: Couldn't connect to any host.  FastAGI failed.

1) Please try to use a subject line that actually provides a subject.

I didnt know what to subject line it, any suggestions, ill edit it?

2) You did not post your installation information. Vicibox? Goautodial? A link to the installation instruction set you used? These are all very important if you are having a problem getting the system online. In fact, without that information it could take weeks to wander step-by-step through "how to get the system working" with back and forth. But knowing where we have started makes a huge difference. Huge.

I mounted the image on a virtual server, aws lightsail and used the "scratch install" instructions included within docs

3) Network topology is important here. Is the IAX2/SIP workstation on the same subnet as the Vicidial server?

no, i am dialing using sip/iax via zoiper from my home laptop

Are you accessing via public or private IP? the server has a static ip assigned that i ssh to.

Is NAT involved? only in the form of "nat=yes" within sip.conf the firewall on the server has all ports udp and tcp open atm, my personal firewall at home is off and there is no firewall software running on the server

4) Why are we discussing a carrier during your login? Telephone company is not related to a login via SIP or IAX2.

i thought possible dialplan error

5) Have you verified that your sip.conf's "externip" value is set to your public IP? yes it is set to my server static ip

6) that portions from my dialplan...

You're all over the place. Perhaps a more linear path would be useful. Slower attempt.

I mounted the image on a virtual server, aws lightsail and used the "scratch install" instructions included within docs

What image? What instructions? What docs?

I strongly suspect you need to start with the PDF installation instructions available on Vicibox.com's site and meant to be used to install the .iso image from that same site. Follow instructions and stop if you have any errors. Bring the error here along with an explanation (page/line instruction?) about what was happening when the error occurred. Not six or seven different issues, just the first error. One step at a time. If you successfully get through the installtion of the .iso from the INstall PDF< then you switch to the Vicidial Manager's Manual.

Start at page one of the Vicidial Manager's Manual (available on EFLO.net). Don't skip anything. Stop when you hit your first error. Once again, bring the error with the page/line/version of the manual and what went wrong along with any relevant configuration information related to the error. (ie: what happened, what you expected to happen ...)

Sigh, i tried the vicibox install with instructions, vicibox9.0.2 the newest one, after the install is said and done, ive tried different ways, both vicibox-install and vicibox-express on a local desktop, neither one installs correctly, it says its finished but yet there is no folder inside /var containing "www" and in turn "html", so at that point i ran install.pl to try to finish the install as it appears not to finish correctly, and apache2 is running yet, connection is refused to get to /vicidial/admin.php or welcome.php and even after changing the permissions via chmod of the entire directory of html, still refused, so im guessing something to do with php7 not installing correctly, but your guess is better than mine.

Where in the instructions do you find "check for /var/www"?

I think you are too "techie" for your own good. These instructions are designed to get a Newbie through the installation process. If you apply your technie brain instead of your "i'm just following instructions in a pdf" brain, you may never get to the end of the road due to all the distractions.

Summary: www is not in VAR in the OpenSuSE continuum. In SuSE the web root is /srv/www/htdocs

Please go back, try again, and don't be smart. Just follow directions without invoking any knowledge outside this install. One page at a time. Stop when you hit an error and bring the error here. Someone will walk you through. Go slow. Have someone else do it without interfering with them if you are unable to turn off your techie brain. I had my eight-year-old do this install a decade ago. It's really designed for "i'm not scared of computers" rather than "I can build a server with my eyes closed".

Happy Hunting!

Had you merely hit google up for "vicidial where is /var/www" you would have had this as your first result:

viewtopic.php?p=62693

either way, there was no web pages being served up....but i will try again, and check if theres something i missed in the 400 page manual i have :p thank you for your time sir

There are only 50-75 in a successful installation. The rest is "oooo!" stuff.

lmao thx bro

also there is a vicibox.log file for the initial install. if there are errors in this log, be sure they do not affect the basic install. ie: if there's an error in the apache install, but apache works ... it may not be worth chasing down. but if there's no web, and there's an error in the apache install, that may be worth investigating.

once you get through the install with the pdf from the vicibox site, switch to the manager's manual and start at page one. it's gotten thousands of newbie users through the installation.

remember that while vicidial has a web GUI, that's only a single portion of Vicidial. it also has "screens" that run (screen -list). about a dozen of them. each of these is a perl script with a task to accomplish. there's also a lot of entries in the cron system for continued operations. and perl runs in "hi-res" timing mode, so it requires every nanosecond (every tick of the CPU's clock). Thus virtual is problematic once you get past one or two agents. works very well for testing and development, but when you hit full power you will not enjoy the experience. not to say "cloud" is bad, just be sure to get a dedicated server in the cloud, not a virtual one. no shared CPU.

Happy Hunting!

once again, i appreciate your time, i am doing an install now on my local desktop, its core2quad 3.33ghz, 8gb ram and 200ssd so there shouldnt be any issues with processing power, thanks for the heads up on the logfile, i do believe i remember seeing the install error over creating it the last time i did it, cuz ive tried 3 times so far, but i will make sure to keep note of it thi run and what the error is when it tries to create it if i remember correctly and let u guys know. getting asterisk running with the correct socks that i choose has never been much of an issue since my venture on this project like 3 weeks ago, (still havent made a successful call) and i mean endless nights of reading and researching, i actually have the 400 page manual....plus when i first started it was issues with centos and then ubuntu, trying to resolve the issues at the base, on their respective forums, then i moved up to asterisk and their forum, so im not a techie as u say, but i have learned how each part operates as ive gone, thru trial and error and research and reading, i have been on the back end of vicidial/goautodial/unidial/vicidialnow in call centers for over 15 years, i can operate the gui portion pretty well, i live in south florida and since ur in orlando, u know how big our call centers get down here in miami and fort lauderdale....ive worked for 2 dozen vacation rooms over the past 20 years, and helped build 2 of the largest in south fl, starting with 4 agents and ending with 150. so im actually taking pleasure in all this learning parts to it, doing it the hard way if u will, because it will make me better at my job.....but last night i decided to give in and just install on my desktop and i havent slept yet, haha, lots of cigs and coffee and soda...anyways ill end my rants, thx again :p

While testing, I HIGHLY recommend virtual. Once you have the virtual perfected for a single test agent, physical server installation is "timely". But the ability to snapshot and restore and reinstall and fastboot ... essential to the playtime involved in learning Vicidial. Of course, if your virtual is on a local network (vSphere is perfect for this), things are simpler. No NAT.

Remember: Anything you do outside the "follow instructions" is likely to be what your problem is. Don't step outside the lines until you're making calls with a fully functional system (inbound, outbound, blended, transfers ...). THEN start enhancing.

Good luck!