[resolved] my demo is not working :(

[emel_punk]

Hi all
ive recently installed vicidial and ive been folowing the instruction on the manual sample version.
but there is a part when im kinda stuck.

i just simply log an agent into
mywebsite /vicidial/welcome.php -> Agent Login
and then log the extension. 10001 + passwd
after that i choose a campaign and my agent info
my phone rings and when i answer it just playback me a recording that says:
im sorry thats not a valid extension, please try again. and my /var/log/asterisk/messages looks like this:

[Nov 29 16:05:54] VERBOSE[23311] logger.c: [Nov 29 16:05:54] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 29 16:05:54] DEBUG[23311] chan_iax2.c: prepending 2 to prefs
[Nov 29 16:05:54] VERBOSE[2592] logger.c: [Nov 29 16:05:54] -- Call accepted by 192.168.0.201 (format gsm)
[Nov 29 16:05:54] VERBOSE[2592] logger.c: [Nov 29 16:05:54] -- Format for call is gsm
[Nov 29 16:05:56] VERBOSE[23311] logger.c: [Nov 29 16:05:56] > Channel IAX2/10001-15791 was answered.
[Nov 29 16:05:56] VERBOSE[23315] logger.c: [Nov 29 16:05:56] == Starting IAX2/10001-15791 at default,s,1 failed so falling back to exten 's'
[Nov 29 16:05:56] VERBOSE[23315] logger.c: [Nov 29 16:05:56] == Starting IAX2/10001-15791 at default,s,1 still failed so falling back to context 'default'
[Nov 29 16:05:56] VERBOSE[23315] logger.c: [Nov 29 16:05:56] -- Sent into invalid extension 's' in context 'default' on IAX2/10001-15791
[Nov 29 16:05:56] VERBOSE[23315] logger.c: [Nov 29 16:05:56] -- Executing [i@default:1] Playback("IAX2/10001-15791", "invalid") in new stack
[Nov 29 16:05:56] VERBOSE[23315] logger.c: [Nov 29 16:05:56] -- <IAX2/10001-15791> Playing 'invalid' (language 'en')
[Nov 29 16:05:58] VERBOSE[23311] logger.c: [Nov 29 16:05:58] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] == Auto fallthrough, channel 'IAX2/10001-15791' status is 'UNKNOWN'
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] -- Executing [h@default:1] DeadAGI("IAX2/10001-15791", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_network: yes
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_network_script: call_log--HVcauses--PRI-----NODEBUG-----0---------------
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_request: agi://127.0.0.1:4577/call_log--HVcauses ... ----------
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_channel: IAX2/10001-15791
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_language: en
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_type: IAX2
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_uniqueid: 1291064754.3
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_callerid: 0000000000
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_calleridname: S101129160554
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_callingpres: 0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_callingani2: 0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_callington: 0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_callingtns: 0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_dnid: unknown
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_rdnis: unknown
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_context: default
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_extension: h
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_priority: 1
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_enhanced: 0.0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >> agi_accountcode:
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] AGI Tx >>
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Nov 29 16:06:00] VERBOSE[23315] logger.c: [Nov 29 16:06:00] -- Hungup 'IAX2/10001-15791'



any helps?

Thanks

[williamconley]

1)

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 Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

2)

ive been folowing the instruction on the manual sample version

Which manual? What Version? What page are you on?

3)

When you set up the carrier, what settings did you give it (list them all, change passwords to PWD and users to USER and subdomain.domain.com to DOMAIN). Be sure to include settings for the account, global string, and dial plan for this carrier.

[emel_punk]

Hello Thanks for you support here is the info. let me know whats left?

1) ive downloaded the server iso from vicidial i dont know the version so this is waht show me my sistem
suse 11.3 installed Linux 2.6.34.4-0.1-default comes with asterisk :1.4.27.

2) manua says
for vicidial release 2.2.0 page 14 and the step 29." You should now be logged in as agent 7777 and your phone should ring to place you into the ViciDial session"

3) carrier settings

ACCOUNT ENTRY
[mudhoney]
disallow=all
allow=ulaw
type=friend
username=username
secret=secret
host=dynamic
dtmfmode=rfc2833
context=trunkinbound

DIALPLAN ENTRY
exten => _9XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXX,2,Dial(newsip:test@DOMAIN:5060/${EXTEN:1},,tTor)
exten => _9XXXXXXX,3,Hangup

REGISTRATION STRING
register => username:password@DOMAIN:5060


SERVERIP
DOMAIN

[emel_punk]

oh and the manual is administration manual

[williamconley]

1) when you log in through ssh or on the console, the system will tell you the version of vicibox (if tht's what you used) (ie: 3.x.x or 2.x.x) in the splash screen.

2) show cli from the attempted agent login moment (not 3000 lines of code, just the moment of login with perhaps 5 lines above and below)

3) account entry "host=dynamic" is not a valid entry, you must check with your provider and get the sip context settings for username/secret/host (dynamic means they will register with you to allow asterisk to set the host ... your provider will NOT be registering with you, it works the other way around, and asterisk certainly cannot Guess the ip address of your provider)

4) you did not post your globals string which should be something akin to:

Code: Select all

TRUNK9=SIP/mudhoney

5) your dialplan entry "exten" (_9XXXXXXX) is valid ONLY if your phone numbers are all 7 digit phone numbers, you have "dial prefix=9" in your campaign, and you have no phone code (or omit phone code checked). What is the telephone number pattern for your area? US Dial Pattern:

Code: Select all

_91NXXNXXXXXX

combined with a campaign dial prefix of "9" and a dial code of "1" and standard 10 digit US phone numbers, of course.

6) your dialplan entry should not contain your sip account information directly, rather it should have your global string variable instead of your direct SIP account information.

Code: Select all

Dial(${TRUNK9}/${EXTEN:1},,tTor)

[emel_punk]

Hi
1) Thank you for installing ViciBox Server v.3.0!
2) thats what i did. that is what appear on /var/log/asterisk/messages when the call is in progress, just that its more detail on the file instead of catch it from the cli. But anyway, here is

Nov 30 17:10:34] -- Format for call is gsm
[Nov 30 17:10:39] > Channel IAX2/10001-14102 was answered
[Nov 30 17:10:39] == Starting IAX2/10001-14102 at default,s,1 failed so falling back to exten 's'
[Nov 30 17:10:39] == Starting IAX2/10001-14102 at default,s,1 still failed so falling back to context 'default'
[Nov 30 17:10:39] -- Sent into invalid extension 's' in context 'default' on IAX2/10001-14102
[Nov 30 17:10:39] -- Executing [i@default:1] Playback("IAX2/10001-14102", "invalid") in new stack
[Nov 30 17:10:39] -- <IAX2/10001-14102> Playing 'invalid' (language 'en')
[Nov 30 17:10:41] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 30 17:10:43] == Auto fallthrough, channel 'IAX2/10001-14102' status is 'UNKNOWN'
[Nov 30 17:10:43] -- Executing [h@default:1] DeadAGI("IAX2/10001-14102", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 30 17:10:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Nov 30 17:10:43] -- Hungup 'IAX2/10001-14102'

3) i dont have a sip provider, what i did was try to create a trunk between another asterisk i have in my lan.

4) TRUNKpixies = IAX2/USERNAME:PASSWORD@IPSERVER:4569

5) yes i have a prefix into my campaign

6) i just did whay the manual says. :s

[williamconley]

2) thats what i did. that is what appear on /var/log/asterisk/messages when the call is in progress, just that its more detail on the file instead of catch it from the cli. But anyway, here is

Nov 30 17:10:34] -- Format for call is gsm
[Nov 30 17:10:39] > Channel IAX2/10001-14102 was answered
[Nov 30 17:10:39] == Starting IAX2/10001-14102 at default,s,1 failed so falling back to exten 's'

You have not successfully added your IAX phone. Try using the pre-installed SIP phone instead (gs102) or use the script which will add 50 sip and 50 iax2 phones for you properly (be sure to run the server ip change script afterwards so they are assigned to your server instead of 10.10.10.15).

3) i dont have a sip provider, what i did was try to create a trunk between another asterisk i have in my lan.

that does not change the fact that you must have a host entry. instead of contacting a carrier and asking what the host entry is, you will have to supply the ip address of your server. this should be much easier, since i presume you KNOW the ip address of the server in question. host cannot be dynamic.

4) TRUNKpixies = IAX2/USERNAME:PASSWORD@IPSERVER:4569

Nope. TRUNKpixies=IAX2/mudhoney (or the name of the IAX2 context you provided in your account entry)

5) yes i have a prefix into my campaign

the dial pattern for your country?

6) i just did whay the manual says. :s

what page of which manual states that you should put the entire carrier registration string into your Dial command instead of a global variable? (Doesn't change the fact that I don't recommend it, I'd just like to see that for myself so I can have a discussion with the publisher of the manual on said topic)

[emel_punk]

This is weird .. ive been using the free manual and none of the things said there works. So i need to prove to my boss that this thing works but .... what tha hell.
I dont know where is the scipt you talking about.

the page it says about the carrier trunk is page 9

2. For this tutorial we will use the following values for the fields on the ADD A NEW CARRIER
form:


Carrier ID: NEWSIP
Carrier Name: ViciDial SIP carrier
Registration String: register => newsip:test@10.10.10.15:5060
<we will leave the Template ID field blank>
Account Entry: (see notes at the end of this tutorial for more information)
[testcarrier]
disallow=all
allow=ulaw
type=friend
username=testcarrier
secret=test
host=dynamic
dtmfmode=rfc2833
context=trunkinbound
Protocol: SIP
<we will leave the Globals String field blank>
Dialplan Entry:
Example value: “SIP/testcarrier”
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(newsip:test@10.10.10.15:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup
Server IP: 10.10.10.15 (set this to your server ip)

i've done the change from dynamic to the server IP but in my other server .. appear an error its:
Peer 'mudhoney' is not dynamic (from 192.168.0.4)
and my iax-vicidial.conf have:
register =>mudhoney:test@192.168.0.2:4569

its looks like is not rewriting my old config 'cause.

[williamconley]

well, then let's go slowly and get it running so you can prove to your boss that these things work. LOL

would you like to start with your Agent phone or your Carrier?

Do you have gs102 (pre-installed) available in your Admin->Phones list?

Do you have a soft phone (X-lite, Zoiper...) installed on a local workstation?

[emel_punk]

Lets begin with the phone, yes i have the gs102 in it and i have xlite, ekiga and kiax.

[williamconley]

cool ... have you successfully registered a phone to the system with the gs102 phone's account?

Code: Select all

sip show registry

[emel_punk]

well not with sip show registry.


but with sip show peers

Name/username Host Dyn Nat ACL Port Status
gs102/gs102 SIPFROMIP D N 5060 OK (23 ms)

[williamconley]

oops, sorry, you're right. i was typing without thinking.

if the correct ip shows under sip show peers you HAVE registered successfully.

next we need to log that phone in as an agent and hear the magical phrase "You are the only person in this conference"

Have you done this yet?

Requires a list, a campaign, and leads in the hopper (OR "allow no leads login" set for the campaign). Using the administrator as the "User" for the login is fine.

[emel_punk]

oh no , thats the stucky part.
when i log on in the page.. my phone rings i answer and play back. IM sorry thats not a valid extension please try again.

[williamconley]

what credentials did you use (gs102 for the phone)?

have you modified gs102 other than the password?

[Kumba]

Sounds to me like you have no good timing source or something else is wrong.

Do you get 99.9+% replies when you run "dahdi_test"?

[emel_punk]

All right . i did it. finally logeed in with gs102, and rings.. but again the same playback... "im sorry thats not a valide station bla bla"

[williamconley]

please post the CLI from that attempt (so we can see the actual message played)

also: when you log in, it assigns you to a conference room (8600051? something like that?) please find it (post it) and post the results from dialing that number from your soft phone. (ie: the number of the conference room)

[emel_punk]

this is the outpute when i login;

[Dec 2 15:25:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 2 15:25:02] > Channel SIP/gs102-00000003 was answered.
[Dec 2 15:25:02] == Starting SIP/gs102-00000003 at default,,1 failed so falling back to exten 's'
[Dec 2 15:25:02] == Starting SIP/gs102-00000003 at default,s,1 still failed so falling back to context 'default'
[Dec 2 15:25:02] -- Sent into invalid extension 's' in context 'default' on SIP/gs102-00000003
[Dec 2 15:25:02] -- Executing [i@default:1] Playback("SIP/gs102-00000003", "invalid") in new stack
[Dec 2 15:25:02] -- <SIP/gs102-00000003> Playing 'invalid' (language 'en')
[Dec 2 15:25:06] == Auto fallthrough, channel 'SIP/gs102-00000003' status is 'UNKNOWN'
[Dec 2 15:25:06] -- Executing [h@default:1] DeadAGI("SIP/gs102-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Dec 2 15:25:06] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 2 15:25:07] == Parsing '/etc/asterisk/manager.conf': [Dec 2 15:25:07] Found


this is the output when y dial 8600051

[Dec 2 15:22:44] -- Executing [8600051@default:1] MeetMe("SIP/gs102-00000002", "8600051|F") in new stack
[Dec 2 15:22:44] == Parsing '/etc/asterisk/meetme.conf': [Dec 2 15:22:44] Found
[Dec 2 15:22:44] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Dec 2 15:22:44] Found
[Dec 2 15:22:44] -- <SIP/gs102-00000002> Playing 'conf-invalid' (language 'en')
[Dec 2 15:22:48] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/gs102-00000002'
[Dec 2 15:22:48] -- Executing [h@default:1] DeadAGI("SIP/gs102-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Dec 2 15:22:48] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

that's interesting. does conference 8600051 exist in your vicidial conferences setup and in your meetme-vicidial.conf?

[emel_punk]

no dude . i does not exist. anywere. in vicidial conferences the maximun room conferences is 8600049.
Should i create it?

[emel_punk]

Plus none of the conference room are not in any meetme* files

[emel_punk]

Oh man am so sorrry it exist 8600051 in vicidial conferences setup.
But i dont see any of them in meetme*.conf

[williamconley]

do you have a meetme-vicidial.conf? is it being recreated when vicidial updates? what is in it?

[emel_punk]

Yeah the file exists but no conference is in it.
i try to put some conf-room manually but it doesnt overwrite the file.
also i try to modify VICIDIAL Conferences parameter Server IP: cause every one of it has a wrong IP server 10.10.10.* . but it doesnt take it anyway.

[williamconley]

run the server update_ip script with that 10.10.10.* value as the "OLD" ip address and your server's actual ip address as new.

[emel_punk]

Dude it worked!!!
Thanks... now i can go on with the manual.
and try to put it up to work

[williamconley]

good post back. now go make some $$