Cannot log into campaign..

[Axiom]

Hi,

I have installed VicidialNOW and setup a campaign and loaded leads.
I can call other exts. and call out to the pstn.
However when I try to log into the campaign, my ext rings and when I answer it I hear " I'm sorry that's not a valid ext."

If you can point me in the direction where to investigate I would appreciate it.
Here is the console log.

Nov 18 09:59:22] -- Unregistered SIP '5000'
[Nov 18 09:59:23] -- Registered SIP 'cc100' at 192.168.1.8 port 47638 expires 3600
[Nov 18 09:59:23] -- Saved useragent "eyeBeam release 1009l stamp 37965" for peer cc100
[Nov 18 09:59:49] == Refreshing DNS lookups.
[Nov 18 10:00:01] == Parsing '/etc/asterisk/manager.conf': [Nov 18 10:00:01] Found
[Nov 18 10:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 18 10:00:01] == Parsing '/etc/asterisk/manager.conf': [Nov 18 10:00:01] Found
[Nov 18 10:00:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 18 10:00:01] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 18 10:00:03] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 18 10:00:06] == Parsing '/etc/asterisk/manager.conf': [Nov 18 10:00:06] Found
[Nov 18 10:00:06] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 18 10:00:06] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 18 10:00:47] == Parsing '/etc/asterisk/manager.conf': [Nov 18 10:00:47] Found
[Nov 18 10:00:47] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 18 10:00:54] > Channel SIP/cc100-09683db0 was answered.
[Nov 18 10:00:54] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 18 10:00:54] == Starting SIP/cc100-09683db0 at default,,1 failed so falling back to exten 's'
[Nov 18 10:00:54] == Starting SIP/cc100-09683db0 at default,s,1 still failed so falling back to context 'default'
[Nov 18 10:00:54] -- Sent into invalid extension 's' in context 'default' on SIP/cc100-09683db0
[Nov 18 10:00:54] -- Executing [i@default:1] Playback("SIP/cc100-09683db0", "invalid") in new stack
[Nov 18 10:00:54] -- <SIP/cc100-09683db0> Playing 'invalid' (language 'en')
[Nov 18 10:00:58] == Auto fallthrough, channel 'SIP/cc100-09683db0' status is 'UNKNOWN'
[Nov 18 10:00:58] -- Executing [h@default:1] DeadAGI("SIP/cc100-09683db0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Nov 18 10:00:58] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[boybawang]

try reloading your setup, go to ADMIN / SERVERS then click on your server then on the last part down you will see REBUILD CONFIG FILES

[Axiom]

I have done as you instructed, rebooted, yet I get the sam results.
I seem that my problem is connecting the open extension to the vicidial conference.
Let me know your thoughts on that.

[mflorell]

What is the output of "show application meetme" in the Asterisk CLI?

[Axiom]

vici*CLI> show application meetme
vici*CLI>
-= Info about application 'MeetMe' =-

[Synopsis]
MeetMe conference bridge

[Description]
MeetMe([confno][,[options][,pin]]): Enters the user into a specified MeetMe
conference. If the conference number is omitted, the user will be prompted
to enter one. User can exit the conference by hangup, or if the 'p' option
is specified, by pressing '#'.
Please note: The Zaptel kernel modules and at least one hardware driver (or ztdummy)
must be present for conferencing to operate properly. In addition, the chan_zap
channel driver must be loaded for the 'i' and 'r' options to operate at all.

The option string may contain zero or more of the following characters:
'a' -- set admin mode
'A' -- set marked mode
'b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
Default: conf-background.agi (Note: This does not work with
non-Zap channels in the same conference)
'c' -- announce user(s) count on joining a conference
'd' -- dynamically add conference
'D' -- dynamically add conference, prompting for a PIN
'e' -- select an empty conference
'E' -- select an empty pinless conference
'F' -- Pass DTMF through the conference.
'i' -- announce user join/leave with review
'I' -- announce user join/leave without review
'l' -- set listen only mode (Listen only, no talking)
'm' -- set initially muted
'M' -- enable music on hold when the conference has a single caller
'o' -- set talker optimization - treats talkers who aren't speaking as
being muted, meaning (a) No encode is done on transmission and
(b) Received audio that is not registered as talking is omitted
causing no buildup in background noise. Note that this option
will be removed in 1.6 and enabled by default.
'p' -- allow user to exit the conference by pressing '#'
'P' -- always prompt for the pin even if it is specified
'q' -- quiet mode (don't play enter/leave sounds)
'r' -- Record conference (records as ${MEETME_RECORDINGFILE}
using format ${MEETME_RECORDINGFORMAT}). Default filename is
meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is
wav.
's' -- Present menu (user or admin) when '*' is received ('send' to menu)
't' -- set talk only mode. (Talk only, no listening)
'T' -- set talker detection (sent to manager interface and meetme list)
'w[(<secs>)]'
-- wait until the marked user enters the conference
'x' -- close the conference when last marked user exits
'X' -- allow user to exit the conference by entering a valid single
digit extension ${MEETME_EXIT_CONTEXT} or the current context
if that variable is not defined.
'1' -- do not play message when first person enters

[mflorell]

Did you change your server IP address after you did your installation?

[Axiom]

Yes, from 192.168.1.2 to 192.168.1.100, I also move the DB server outboard as well, upgraded to Asterisk 1.4.all else work, except logging into the campaign

[mflorell]

Did you run the update_server_ip script?