vicidial.org

Hi,

just a newbie here.
i was ale to setup vicibox redux 4.0.3; Vicidial VERSION: 2.8-407a BUILD: 130709-1350 from; Asterisk 1.4.44 from scratch. I'm also using eyebeam softphone 1.1 3007n stamp 17816.
I was able to create users, campaign, phones, carrier, and able to load leads.
I was able to login as an agent successfully on vicidial and connect to my eyebeam softphone. but when i clicked on "dial next number on vicidial, I'm getting an error msg:" i am sorry, that's not a valid extension. pls try again".
Don't know what to do with this...
Please help...

Thanks,

PS:
below is the carrier setup and CLI msgs:

CARRIER SETUP

Carrier ID

GETONSIP

Carrier Name

getonsip

Registration String:

register =>

username:password@69.57.179.234/username

Account Entry

[getonsip]
type=peer
username=username
secret=password
host=69.57.179.234 (sip host's IP address)
dtmfmode=rfc2833
context=default
insecure=very
disallow=all
canreinvite=no
allow=g729
qualify=1000
nat=yes

Globals String:

GETONSIP= SIP/getonsip

Dialplan Entry

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,1,Dial

(getonsip@69.57.179.234:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


CLI MSG ON ASTERISK

[Jul 17 03:17:16] == Parsing '/etc/asterisk/manager.conf': [Jul 17 03:17:16] Found
[Jul 17 03:17:16] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 03:17:16] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-2bdd,2",

"8600051|F") in new stack
[Jul 17 03:17:16] > Channel Local/8600051@default-2bdd,1 was answered.
[Jul 17 03:17:16] == Starting Local/8600051@default-2bdd,1 at default,93525614878,1 failed so

falling back to exten 's'
[Jul 17 03:17:16] == Starting Local/8600051@default-2bdd,1 at default,s,1 still failed so falling back

to context 'default'
[Jul 17 03:17:16] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-

2bdd,1
[Jul 17 03:17:16] -- Executing [i@default:1] Playback("Local/8600051@default-2bdd,1", "invalid")

in new stack
[Jul 17 03:17:16] -- <Local/8600051@default-2bdd,1> Playing 'invalid' (language 'en')
[Jul 17 03:17:16] WARNING[18937]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 03:17:16] WARNING[18937]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 03:17:18] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 03:17:20] == Auto fallthrough, channel 'Local/8600051@default-2bdd,1' status is

'UNKNOWN'
[Jul 17 03:17:20] -- Executing [h@default:1] DeadAGI("Local/8600051@default-2bdd,1",

"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 03:17:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----

NODEBUG-----0--------------- completed, returning 0
[Jul 17 03:17:20] == Spawn extension (default, 8600051, 1) exited non-zero on

'Local/8600051@default-2bdd,2'
[Jul 17 03:17:20] -- Executing [h@default:1] DeadAGI("Local/8600051@default-2bdd,2",

"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 03:17:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----

NODEBUG-----0--------------- completed, returning 0
[Jul 17 03:17:25] NOTICE[2692]: chan_sip.c:8434 sip_reg_timeout: -- Registration for

'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #36)
zerone*CLI>

Its because the dialed number 93525614878 does not match _91NXXNXXXXXX
you might want to try something like _9XXXXXXXXXX instead in the dialplan

also in the Dial function:
Dial(getonsip@69.57.179.234:5060/${EXTEN:2},,tTor)
you should edit that to be:
Dial(${GETONSIP}/${EXTEN:2},,tTor)

And ${EXTEN:2} cuts away the first two digits in the phone number, so 525614878 will be the number sent to the carrier, change to :1 if you only want to get rid of the first digit

Thanks for the reply DomeDan,

I'm gonna try your advice and will keep you posted

Thanks very much

Hi domeDan,

I've tried your advice, but I'm still the same error msg

DIAL PLAN ENTRY

exten => _91NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXXXXXXXX,1,Dial(${GETONSIP}/${EXTEN:2},,tTor)
exten => _91NXXXXXXXXX,3,Hangup

CLI NOTES

[Jul 17 20:54:31] == Parsing '/etc/asterisk/manager.conf': [Jul 17 20:54:31] Found
[Jul 17 20:54:31] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 20:54:35] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #155)
[Jul 17 20:54:40] > Channel SIP/getonsip01-00000002 was answered.
[Jul 17 20:54:40] -- Executing [8600051@default:1] MeetMe("SIP/getonsip01-00000002", "8600051|F") in new stack
[Jul 17 20:54:40] == Parsing '/etc/asterisk/meetme.conf': [Jul 17 20:54:40] Found
[Jul 17 20:54:40] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Jul 17 20:54:40] Found
[Jul 17 20:54:40] -- Created MeetMe conference 1023 for conference '8600051'
[Jul 17 20:54:40] -- <SIP/getonsip01-00000002> Playing 'conf-onlyperson' (language 'en')
[Jul 17 20:54:40] == Refreshing DNS lookups.
[Jul 17 20:54:42] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 20:54:57] == Parsing '/etc/asterisk/manager.conf': [Jul 17 20:54:57] Found
[Jul 17 20:54:57] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 20:54:57] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-262a,2", "8600051|F") in new stack
[Jul 17 20:54:57] > Channel Local/8600051@default-262a,1 was answered.
[Jul 17 20:54:57] -- Executing [913522057447@default:1] AGI("Local/8600051@default-262a,1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 17 20:54:57] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 17 20:54:57] == Auto fallthrough, channel 'Local/8600051@default-262a,1' status is 'UNKNOWN'
[Jul 17 20:54:57] -- Executing [h@default:1] DeadAGI("Local/8600051@default-262a,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 20:54:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 20:54:57] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-262a,2'
[Jul 17 20:54:57] -- Executing [h@default:1] DeadAGI("Local/8600051@default-262a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 20:54:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 20:54:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 20:55:00] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #156)
[Jul 17 20:55:01] == Parsing '/etc/asterisk/manager.conf': [Jul 17 20:55:01] Found
[Jul 17 20:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 20:55:01] == Parsing '/etc/asterisk/manager.conf': [Jul 17 20:55:01] Found
[Jul 17 20:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 20:55:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 20:55:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 20:55:06] == Parsing '/etc/asterisk/manager.conf': [Jul 17 20:55:06] Found
[Jul 17 20:55:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 20:55:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 20:55:20] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #157)
zerone*CLI>

also tried to change the dial prefix from "9" to "X" in the campaign using the same dial plan. the same error msg. here's the CLI Notes for this changes:

[Jul 17 21:06:14] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:06:14] Found
[Jul 17 21:06:14] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:06:14] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-41a6,2", "8600051|F") in new stack
[Jul 17 21:06:14] > Channel Local/8600051@default-41a6,1 was answered.
[Jul 17 21:06:14] == Starting Local/8600051@default-41a6,1 at default,13527534110,1 failed so falling back to exten 's'
[Jul 17 21:06:14] == Starting Local/8600051@default-41a6,1 at default,s,1 still failed so falling back to context 'default'
[Jul 17 21:06:14] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-41a6,1
[Jul 17 21:06:14] -- Executing [i@default:1] Playback("Local/8600051@default-41a6,1", "invalid") in new stack
[Jul 17 21:06:14] -- <Local/8600051@default-41a6,1> Playing 'invalid' (language 'en')
[Jul 17 21:06:14] WARNING[10163]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 21:06:14] WARNING[10163]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 21:06:16] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:06:18] == Auto fallthrough, channel 'Local/8600051@default-41a6,1' status is 'UNKNOWN'
[Jul 17 21:06:18] -- Executing [h@default:1] DeadAGI("Local/8600051@default-41a6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 21:06:18] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:06:18] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-41a6,2'
[Jul 17 21:06:18] -- Executing [h@default:1] DeadAGI("Local/8600051@default-41a6,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 21:06:18] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:06:19] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #189)
[Jul 17 21:06:40] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #190)
zerone*CLI>

I noticed that the tel number in the previous CLI is 13527534110
so I tried to tried "omit phone code=Y" and "dial prefix=X" in the campaign with this dial plan

exten => _91NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXXXXXXXX,1,Dial(${GETONSIP}/${EXTEN:1},,tTor)
exten => _91NXXXXXXXXX,3,Hangup

CLI NOTES

[Jul 17 21:13:58] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:13:58] Found
[Jul 17 21:13:58] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:13:58] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-2c2f,2", "8600051|F") in new stack
[Jul 17 21:13:58] > Channel Local/8600051@default-2c2f,1 was answered.
[Jul 17 21:13:58] == Starting Local/8600051@default-2c2f,1 at default,3523506700,1 failed so falling back to exten 's'
[Jul 17 21:13:58] == Starting Local/8600051@default-2c2f,1 at default,s,1 still failed so falling back to context 'default'
[Jul 17 21:13:58] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-2c2f,1
[Jul 17 21:13:58] -- Executing [i@default:1] Playback("Local/8600051@default-2c2f,1", "invalid") in new stack
[Jul 17 21:13:58] -- <Local/8600051@default-2c2f,1> Playing 'invalid' (language 'en')
[Jul 17 21:13:58] WARNING[11022]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 21:13:58] WARNING[11022]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jul 17 21:13:59] NOTICE[2577]: chan_sip.c:17722 do_monitor: Disconnecting call 'SIP/agent01-00000003' for lack of RTP activity in 61 seconds
[Jul 17 21:13:59] -- Hungup 'DAHDI/pseudo-1125503318'
[Jul 17 21:13:59] == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/agent01-00000003'
[Jul 17 21:13:59] -- Executing [h@default:1] DeadAGI("SIP/agent01-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 17 21:13:59] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:14:00] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:01] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:14:01] Found
[Jul 17 21:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:14:01] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:14:01] Found
[Jul 17 21:14:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:14:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:02] == Auto fallthrough, channel 'Local/8600051@default-2c2f,1' status is 'UNKNOWN'
[Jul 17 21:14:02] -- Executing [h@default:1] DeadAGI("Local/8600051@default-2c2f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 21:14:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:14:02] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-2c2f,2'
[Jul 17 21:14:02] -- Executing [h@default:1] DeadAGI("Local/8600051@default-2c2f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 17 21:14:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:14:06] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:14:06] Found
[Jul 17 21:14:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:14:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:11] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #212)
[Jul 17 21:14:22] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:14:22] Found
[Jul 17 21:14:22] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:14:22] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:14:22] Found
[Jul 17 21:14:22] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:14:22] -- Executing [55558600052@default:1] MeetMeAdmin("Local/55558600052@default-3fc2,2", "8600052|K") in new stack
[Jul 17 21:14:22] WARNING[11085]: app_meetme.c:3145 admin_exec: Conference number '8600052' not found!
[Jul 17 21:14:22] -- Executing [55558600052@default:2] Hangup("Local/55558600052@default-3fc2,2", "") in new stack
[Jul 17 21:14:22] == Spawn extension (default, 55558600052, 2) exited non-zero on 'Local/55558600052@default-3fc2,2'
[Jul 17 21:14:22] -- Executing [h@default:1] DeadAGI("Local/55558600052@default-3fc2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 17 21:14:22] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 17 21:14:24] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:24] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:14:24] NOTICE[2577]: chan_sip.c:14057 handle_response_peerpoke: Peer '100' is now Reachable. (17ms / 2000ms)
[Jul 17 21:14:32] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #213)
[Jul 17 21:14:40] == Refreshing DNS lookups.
[Jul 17 21:14:52] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #214)
[Jul 17 21:15:02] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:15:02] Found
[Jul 17 21:15:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:15:02] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:15:02] Found
[Jul 17 21:15:02] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:15:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 17 21:15:07] == Parsing '/etc/asterisk/manager.conf': [Jul 17 21:15:07] Found
[Jul 17 21:15:07] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 17 21:15:07] == Manager 'sendcron' logged off from 127.0.0.1
zerone*CLI>

after trying all of those, I'm still getting "iam sorry, that's not a valid extension. pls try again" error msg.
Pls HELP..
Thanks

I see a lot of:
[Jul 17 20:54:35] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #155)

fix that first, figure out why it times out

and change back to 9 dial prefix and stuff

DomeDan wrote:I see a lot of:
[Jul 17 20:54:35] NOTICE[2577]: chan_sip.c:8434 sip_reg_timeout: -- Registration for 'zer1getonsip@69.57.179.234' timed out, trying again (Attempt #155)

fix that first, figure out why it times out

and change back to 9 dial prefix and stuff

thanks DomeDan,

sorry for the ignorance, this is my first time to setup such a thing..
not really sure how to fix it.

do you think it has something to do with my free sip account?

can you recommend a free sip provider for testing purposes?

can Vonage be used for vicidial, if yes how to configure it?

Thanks for your help, really appreciate it

No problem, everyone is a beginner once.

Try to use the free account in a softphone and verify that it wont work first.

And I have no experience with free voip accounts,
or hmm, remember I've seen something like that on a gardo site, ah yeah: http://justgovoip.com/ 60 minutes

oh, that's great "m gonna try that site and see if i can signup for a free working account

Thanks for your patience with me...

I'll keep you posted.

Thanks again

hi DomeDan,

sorry for the very late update...

we tried to get a voip provider, we're currently testing their free demo account before we proceed on getting their service.
they provided their dial plan, but when we tried to their number, according to them its ringing on their end but when they answer the call, we could not hear anything on both my end and theirs (provider).

also, we tried calling our vonage line thru the sip phone, the sip phone ring once and then immediately connects the call, but our vonage phone did not receive the call, it did not even ring.

what would you think the problem was?
can you recommend any voip provider in the US?


Thanks

the least expensive "startup" provider I've seen (that works) is VOIPStreet. $10 on the account will get you going. I think they used to give a free $1 worth of calls, but I think they stopped that.

a simpler solution would be to use an in-house soft phone as the fake phone and route all calls to that phone. then you can fake sales calls to that sip phone forever for free.

williamconley wrote:the least expensive "startup" provider I've seen (that works) is VOIPStreet. $10 on the account will get you going. I think they used to give a free $1 worth of calls, but I think they stopped that.

a simpler solution would be to use an in-house soft phone as the fake phone and route all calls to that phone. then you can fake sales calls to that sip phone forever for free.

Hi williamconley,

I'm gonna check VOIPStreet for possible provider.
It sounds interesting for your simpler solution of a free calls. Just a newbie on this thing, if you could please elaborate on this and if you could please teach me how to do it. I would like to learn this, even just for testing our dialer.

Thank you very much

Code: Select all

exten => _01NXXNXXXXXX,1,Set(CALLERID(num)=5555551212)
exten => _01NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _01NXXNXXXXXX,n,Dial(SIP/cc150,,tTor)
exten => _01NXXNXXXXXX,n,Hangup

this will call standard phone "cc150" regardless of the number that was in the list. So phone cc150 becomes the "perpetual prospect". If you have that as a 2nd VOIP phone in your system you can talk to yourself all day long for free.

williamconley wrote:

Code: Select all

exten => _01NXXNXXXXXX,1,Set(CALLERID(num)=5555551212)
exten => _01NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _01NXXNXXXXXX,n,Dial(SIP/cc150,,tTor)
exten => _01NXXNXXXXXX,n,Hangup

this will call standard phone "cc150" regardless of the number that was in the list. So phone cc150 becomes the "perpetual prospect". If you have that as a 2nd VOIP phone in your system you can talk to yourself all day long for free.

Thanks for the reply williamconley,

Sorry for the ignorance, just a newbie.
If i got it correctly, this is the "dial plan entry" for the carrier setup of the dialer.

exten => _01NXXNXXXXXX,1,Set(CALLERID(num)=5555551212)
exten => _01NXXNXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _01NXXNXXXXXX,n,Dial(SIP/cc150,,tTor)
exten => _01NXXNXXXXXX,n,Hangup[/code]

if the above is the "dial plan entry"
what would be the "account entry", "registration string", and "global string"

im a bit confuse about this:
- if you could please explain this further
- what do you mean by standard phone cc150?

this will call standard phone "cc150" regardless of the number that was in the list. So phone cc150 becomes the "perpetual prospect". If you have that as a 2nd VOIP phone in your system you can talk to yourself all day long for free.


I really want to learn the ins and outs of the VOIP.

Thank you very much for your patience

no need for a global string or account entry.

but you will need to create a phone extension "cc150" (or change that to a phone you actually have).

there is a sql script in the "extras" folder that will add 50 SIP and IAX phones ... cc150 is one of them. (be sure to change the IP of cc150 to your server ... those sample phones will all have the wrong IP on them at first).

You set yourself up as an agent with a normal agent phone. Perhaps using "Zoiper". Then you also register a phone (perhaps with "X-lite" or 3CX) to cc150. Then this method will cause cc150 to ring every time the system tries to call anyone for any reason when configured as described. So no matter what the phone number is in the List ... cc150 will ring. And you can simulate any sort of response you want and "see how it works". No money involved. No actual carrier. Just cc150.