vicidial.org

I get the above error message if the agent login. Someone who can help?


Connected to Asterisk 1.2.27 currently running on vp-dialer (pid = 1710)
Verbosity is at least 21
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc100-09032658 was answered.
-- Executing MeetMe("SIP/cc100-09032658", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Feb 7 20:09:32 WARNING[11513]: chan_zap.c:1064 zt_open: Unable to open '/dev/za p/pseudo': Permission denied
Feb 7 20:09:32 ERROR[11513]: chan_zap.c:7570 chandup: Unable to dup channel: Pe rmission denied
Feb 7 20:09:32 WARNING[11513]: app_meetme.c:480 build_conf: Unable to open pseu do channel - trying device
Feb 7 20:09:32 WARNING[11513]: app_meetme.c:483 build_conf: Unable to open pseu do device
-- Playing 'conf-invalid' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-0903265 8'
-- Executing DeadAGI("SIP/cc100-09032658", "agi://127.0.0.1:4577/call_log--H Vcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6--------------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vp-dialer*CLI>

Hardware configuration and software versions?

I see zap in there, do you have digium hardware in your box?

Have you tested the ability to dial out straight from an extension?

Have you tested the ability to dial into a conference directly (do conferences work)?

I get the same response as original poster.

I was not involved with install for my co, tho, and do not know the hardware that is involved. I do, however, have ssh access to the server install.

I know that the software version is 2.0.4.

Please, is there a way for me to determine the hardware present? Also, how can I find a similar log in my install as the original poster used? I will reply with the specs as soon as is possible. Thank you.

asterisk -R (shows asterisk command line)

at asterisk command line:

set verbose 20 (shows server dial plan in operation)

set verbose 0 (shuts off that view so you can issue other commands without all that mess on the screen while you're working)

you really need to go to some sites like http://www.voip-info.org/wiki/view/Asterisk to get acquainted with your basic server functions before trying to administer vicidial. there's a lot of information to be had.

Thank you for your response.

And, at a rough guess, you're going to find that you don't have {any,enough} 'Vicidial Conferences' set up in admin -- not 'Conferences'; they're different.

Thank you, Baylink.

I will look into that as well.

Hi all,

Same error as OP and jnewgent.

We are running asterisk 1.2.27, a vicidialnow deployment on a VPS.

We had the provider compile ztdummy, but the error persists.

I can verify that we can login to the agent extension using a softphone, and make outbound calls.


Thanks in advance

Would VPS be some kind of virtual server?

If so, we would recommend getting a dedicated machine.

Thanks for the reply.

Yes, VPS would in fact be a virtual server.

Actually, there seems to have been an error the first time we requested the Zaptel dummy to be compiled. So the service provider did it over and its working fine now.

So for everyone else having the same problem, i would recommend double checking ztdummy.

Regards,
PrinzPH