Hopper not Loading Leads - no server ip in Debug

[chrisnetronix]

ViciBox_Redux.i686-2.0.2 from .iso | Vicidial 2.2.1-237 Build 100510-2015 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

I deleted the campaign and recreated it, recreated the lead list, which only has 10 leads.

its on 24 hour dial plan

and still no leads in hopper

the dial status is at new

Dial Status 1: NEW - New Lead

This campaign has 1 active lists and 0 inactive lists

|| This campaign has 10 leads to be dialed in those lists - HIDE

This campaign has 0 leads in the dial hopper


even if i change the dial status to something else it still loads no leads in the hopper


useing:

/usr/share/astguiclient/AST_VDhopper.pl --debugX

it says:


----- DEBUG -----


----- SUPER DEBUG -----

Server IP does not have an entry in the servers table


Also I changed my dial prefix to x because I do not use 9 to dial out

I find this strange because if the debug for the hopper says that why is there info for the server in the mysql db under the servers table


What should I do?

[williamconley]

Server IP does not have an entry in the servers table

Try ADMIN->Servers and see if the server has an IP address ... if it does not, you should put one in. and then we have to find out WHY it did not have one. and if it is missing anywhere else.

[chrisnetronix]

servers link has testast with my server ip and it also shows in mysql under the servers table, where else does it need to be?

[williamconley]

check the /usr/share/astguiclient folder for the admin update server ip script. it is designed to locate and set the server ip everywhere it needs to be changed when it changes.

i'd be nervous, though if any of them are "wrong", that doesn't generally happen unless there is a flaw with the installation (or you manually tried to change the ip address without that script).

since you installed from .iso ... that would make me quite nervous (i'd consider a reinstall to be sure you won't spend the next week dealing with other oddities that occurred when that one did).

on the other hand, if you DID change the ip address of the server without using the ip update script, use it now (it will need old/new ip addresses, and it will seek out the old one and update it to the new one wherever you missed one EXCEPT the externip in sip.conf if you've set that manually)

[chrisnetronix]

the script worked now the hopper is loading leads, now i cant get the user to login correclty via x-lite or agent interface>

cli:

NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found
[Aug 24 00:02:35] NOTICE[3703]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@ 66.71.249.170>' failed for '71.203.248.167' - No matching peer found

[chrisnetronix]

I went to phones and put in the server ip and username and passoerd of the 7777 agent and now it shows

there are no available sessions

cli

[Aug 24 00:52:16] NOTICE[3936]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@66.71.249.170>' failed for '71.203.248.167' - Wrong password


the above is from trying to login via xlite

if i try to login via vicidial agent i get in cli:

[Aug 24 00:53:52] NOTICE[7503]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/7777

[chrisnetronix]

Heres the sip debug cli:


SIP Debugging re-enabled
[Aug 24 02:14:19]
<--- SIP read from 70.167.153.130:5060 --->
OPTIONS sip:66.71.249.170:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:70.167.153.130;lr>
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKd14f.8ef2bbd501a51aaf2c3592ef000f9621.0
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
Route: <sip:70.167.153.130;lr;received="sip:66.71.249.170:5060">
From: sip:ping@invalid;tag=f549cba6
To: sip:66.71.249.170:5060
Call-ID: 887d2e83-118d3166-22d46a4@216.115.69.131
CSeq: 1 OPTIONS
Content-Length: 0


<------------->
[Aug 24 02:14:19] --- (11 headers 0 lines) ---
[Aug 24 02:14:19] Looking for s in default (domain 66.71.249.170)
[Aug 24 02:14:19]
<--- Transmitting (NAT) to 70.167.153.130:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bKd14f.8ef2bbd501a51aaf2c3592ef000f9621.0;received=70.167.153.130
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
From: sip:ping@invalid;tag=f549cba6
To: sip:66.71.249.170:5060;tag=as7f13dfe6
Call-ID: 887d2e83-118d3166-22d46a4@216.115.69.131
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[Aug 24 02:14:19] Scheduling destruction of SIP dialog '887d2e83-118d3166-22d46a4@216.115.69.131' in 32000 ms (Method: OPTIONS)
[Aug 24 02:14:24] Really destroying SIP dialog 'MTQwNmFjMWRjZGNiNDY5ZDM3Mzk3ZjI2NThlN2UxNTg.' Method: REGISTER

[chrisnetronix]

now when trying to login via xlite i get:

[Aug 24 02:41:00] NOTICE[3936]: chan_sip.c:16378 handle_request_register: Registration from '"7777"<sip:7777@66.71.249.170>' failed for '71.203.248.167' - No matching peer found

[williamconley]

when you add a new "phone" under admin, 60 seconds later vicidial should "rewrite" the sip-vicidial.conf file (with the new entries in it). if it does not, you need to verify that admin->servers entry for this server has generate conf files selected.

[chrisnetronix]

this is what the vicidail sip conf says:

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
register => 14512814:xxxxxxxr@sip.flowroute.com

; VICIDIAL Carrier: Flowroute - Flowroute
; test
[flowroute]
type=friend
secret=xxxxxxxx
username=14512814
host=sip.flowroute.com
dtmfmode=inband
rfc2833compensate=yes
context=inbound
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.flowroute.com



[102]
username=102
secret=test
accountcode=102
callerid="Test Admin Phone" <>
mailbox=102
context=default
type=friend
host=dynamic


; END OF FILE



this is what it says in the phone area under 102

Phone extension:102
Dial Plan Number:102
Voicemail Box:102
Server IP: 66.71.249.172
Server IP:7777
Password:7777
Status:ADMIN
Active Account:y

now in users i have an agent made username 7777 and pass 7777

[williamconley]

and which manual are you following for your tutorial on how to set this up?

[chrisnetronix]

managers manual, 100 bucks

[williamconley]

what page are you on?

[chrisnetronix]

31. You will then need to enter in your username(7777) and password(1234) and select your
campaign(TESTCAMP) then click submit.

page 9

but i made the password 7777 instead of 1234

[chrisnetronix]

what am I doing wrong

[williamconley]

that's not the page number. you trying to confuse the old guy?

it would seem that you did not set up your phone properly first.

the line above that says "NOTE: if you need to set up a phone on your system, see Tutorial A above". You'll need to go back to tutorial A and get that phone set up first.

To be sure it is set up properly, it will need to "register".

The phone you are using should have an indication of whether it is registered or not, and the asterisk command "sip show peers" will also indicate if it has registered (it will have an IP address if it is registered).

[chrisnetronix]

I read all through tutorial A, and it says this about setting up phones at the beginning:

This tutorial assumes that your System Administrator has already set up
your phone entries for your agents' phones.

on page 66 is:

ADMIN - PHONES
Add a New Phone


but it just shows definitions to the fields, it does not have a step by step to setup, I just need to know what to put in the fields correctly

[williamconley]

So page 11 does not have "A. Add a new SIP or IAX phone to the system" and shortly thereafter "2. For this tutorial we will use the following values for the fields on the ADD A NEW PHONE form:"?

Perhaps when I said "which manual are you using" I should have asked "which version of the manager's manual are you using"?

[chrisnetronix]

I had 3 managers manuals and i was reading an old one, now I am reading version 220.pdf

5/10/2010

I see where you are talking about now, now I can bet it to login to xlite correctly but when I go to the vicidial agent login it tells me:

Sorry, there are no available sessions

and cli says:

[Aug 24 21:23:04] NOTICE[3751]: chan_sip.c:3115 auto_congest: Auto-congesting SIP/102-0000000c
[Aug 24 21:23:04] > Channel SIP/102-0000000c was never answered.

[chrisnetronix]

I can get xlite to login but when I log in on the agents login area i get this:

Sorry, there are no available sessions

I dont know what to do now?

[chrisnetronix]

now make note, this is a colocacted server this is on and I am modifying via ssh, and logging in via x lite, and logging in to the agent interface remotely

yet when logging in into the agent interface i get:

Sorry, there are no available sessions

cli

[Aug 24 22:31:38] NOTICE[21567]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/102

[williamconley]

The asterisk system could not reach the sip phone (lag may have been too long) so it "marked" the phone inoperable (auto-congesting) which causes asterisk to refuse to even attempt to dial the sip phone.

turn off "qualify" or improve the lag time (or open the firewall port?) so asterisk will dial the sip phone and THAT problem should be resolved and you can work on others

[THUFIR]

The asterisk system could not reach the sip phone (lag may have been too long) so it "marked" the phone inoperable (auto-congesting) which causes asterisk to refuse to even attempt to dial the sip phone.

turn off "qualify" or improve the lag time (or open the firewall port?) so asterisk will dial the sip phone and THAT problem should be resolved and you can work on others

Very interesting. It's entirely possible to use Asterisk completely "offline", though; just on a LAN. Qualify is an Asterisk setting, not ViciDial per se?