You have been disconnected

[apb1963]

Just installed (first time!) GoAutoDial 3.3 on a centOS 5.1 box. No extra hardware, the only other thing installed is phplist. Dedicated server, not shared - but hosted elsewhere.

Been all through google, vicidial, goautodial and asterisk forums. Posted at goautodial forums, but no response so far.

When I login as an agent the softphone does not receive the initial incoming call and within a few seconds it says "You have been disconnected".

I didn't realize there was a Getting Started guide until after I'd already installed goautodial, had a softphone registered and tried to login. At that point I went back and read the guide. I did deviate a bit from the guide, in that I didn't create a new campaign, I used the test campaign, and didn't use the 8001 account, I used 8007. I have 2 leads in the hopper (correct since it's just a test list of 3, and I used one up trying to manually dial out before I realized I had never received the initial call putting me in a conference. I know my carrier setup probably isn't right, but I presume that's unrelated).

The softphone is registered, this is confirmed by asterisk CLI. I opened the firewall for testing, it did not help.

Then I noticed the /etc/asterisk/meetme.conf file had the following:
~~~~~~~~~~~~~~~~
[rooms]
conf => 8600
conf => 8601,1234

#include meetme-vicidial.conf
~~~~~~~~~~~~~~~~
I added "conf => 8607" since I login as agent007... (I reloaded asterisk) that didn't help, so I took it back out.

The only thing interesting in the asterisk log file is:

[Sep 9 01:16:05] NOTICE2619 channel.c: Unable to request channel SIP/8001

I ran the update_server_ip command although IP hasn't changed. I have alaw, ulaw, gsm and more in softphone. I am using jitsi.

I notice that the vicidial_conferences table shows:

conf_exten server_ip extension leave_3way leave_3way_datetime

8600051 999.999.999.999 SIP/8001 0 NULL
8600052 999.999.999.999 0 NULL
8600053 999.999.999.999 0 NULL
(249 rows)

(IP address redacted of course). Note the extension, SIP/8001. Since I don't know what this table is supposed to look like, I don't know if that's normal?

From the phones table:

Full Texts extension dialplan_number voicemail_id phone_ip computer_ip server_ip login pass status active
Edit Delete 8001 8001 8001 192.168.1.2 999.999.999.999 8001 goautodial ACTIVE Y
Edit Delete 8002 8002 8002 999.999.999.999 8002 goautodial ACTIVE Y
Edit Delete 8003 8003 8003 999.999.999.999 8003 goautodial ACTIVE Y
Edit Delete 8004 8004 8004 999.999.999.999 8004 goautodial ACTIVE Y
Edit Delete 8005 8005 8005 999.999.999.999 8005 goautodial ACTIVE Y
Edit Delete 8006 8006 8006 999.999.999.999 8006 goautodial ACTIVE Y
Edit Delete 8007 8007 8007 999.999.999.999 8007 goautodial ACTIVE Y

The server IP is correct, but I don't know where that computer IP of 192.168.1.2 comes from or if it matters. I don't know if any of this is relevant to my problem.

Since I didn't get a response at the goautodial forums, I decided to do a little experimenting. I cleared out the extraneous data in the two tables mentioned above, and now it put the proper IP address in the 8001 row of the phones table. While that's encouraging, it didn't resolve the issue. Plus it brings up the question of why it's putting my info in that row, when I'm logging in as agent007, NOT agent001. Although it put the proper IP in the table, should it be the PHONE IP and not the COMPUTER IP?

In any event, I'm now getting some additional details in the asterisk log file:

[Sep 9 19:50:00] NOTICE17281 chan_sip.c: Failed to authenticate device 200<sip:200@serverIPredacted>;tag=0ff90fe6 <<< I presume this is a break-in attempt.
[Sep 9 19:50:01] VERBOSE29356 manager.c: [Sep 9 19:50:01] Manager 'sendcron' logged on from 127.0.0.1 (lots of these, I deleted since they don't add any value to this discussion).

[Sep 9 19:50:44] VERBOSE[29461] pbx.c: [Sep 9 19:50:44] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000005;2", "8600051,K") in new stack
[Sep 9 19:50:44] WARNING[29461] app_meetme.c: Conference number '8600051' not found!
[Sep 9 19:50:44] VERBOSE[29461] pbx.c: [Sep 9 19:50:44] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000005;2", "") in new stack
[Sep 9 19:50:44] VERBOSE[29461] pbx.c: [Sep 9 19:50:44] Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000005;2'
[Sep 9 19:50:44] VERBOSE29461 pbx.c: [Sep 9 19:50:44] -- Executing [h@default:1] AGI in new stack
[Sep 9 19:50:44] VERBOSE29461 res_agi.c: [Sep 9 19:50:44] -- <Local/55558600051@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Hopefully this is enough to generate some ideas from someone. I did notice that in another thread that someone had the identical problem, and he said there were lines missing from one of the .conf files (extensions.conf I think it was?). Unfortunately, he failed to provide sufficient information to enable fixing the problem. Maybe someone else knows what those missing lines are?

Also, I saw the various threads regarding ztdummy or dahdi_dummy being needed:

dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.991% 99.987% 99.989% 99.991% 99.990% 99.990% 99.990% 99.990%
99.991% 99.989% 99.989% 99.991% 99.990% 99.912% 99.991% 99.994%
99.987% 99.987% 99.990% 99.992% 99.990% 99.991% 99.990% 99.990%
--- Results after 24 passes ---
Best: 99.994% -- Worst: 99.912% -- Average: 99.986732%
Cummulative Accuracy (not per pass): 99.994

So it looks like i'm good there. At this point, I'm stumped. Any thoughts?

Thank you in advance!

[apb1963]

On a whim, I decided to take a look at "screen".

8559.goautodial_d (Detached)

I attached that process, logged in, and it made the initial call and put me in the conference!!! The question now is, why would I have to do that and how to fix so that I don't?

I should note that the only thing in that screen was:

sh: asterisk: command not found
sh: asterisk: command not found

This _before_ I logged in.

Thank you in advance.