Autodialing not working properly.

[riftdawg]

Hi, I upgraded to 2.04rc2 and suddenly autodialing doesn't work. I can see all the leads in the hopper, I have vdad set to 8364 and my remote agent is setup and active and at ext 8320 (where my broadcast is). Everything is active, I even switched it to 24hr call times, and I dont see any calls at all going through on the asterisk CLI.

[riftdawg]

Should mention that vicidial_auto_calls is empty as well.

And running autodial.pl script in debug gives me the following:

loop counter: |5759995|
TIME DEBUG: -5|-4|1| GMT: 02:03
2008-07-01 22:03:37|LIVE AGENTS LOGGED IN: 0 ACTIVE CALLS: 0|
2008-07-01 22:03:37| : agents: 1 dial_level: 0|
2008-07-01 22:03:37| : Calls to place: 0 (0 - 0) 0 |
2008-07-01 22:03:37|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2008-07-01 22:03:37|LOCAL TRUNK SHORTAGE: 0|0 (0 - 96)|
2008-07-01 22:03:37| : CALLING|
TIME DEBUG: -5|-4|1| GMT: 02:03
2008-07-01 22:03:37|| lagged call vla agent PAUSED 0E0|20080701220307|20080701220327|20080701220337||
2008-07-01 22:03:37|| lagged call vac agent DELETED 0E0|2008-07-01 22:01:37|

Over and over... not sure why it is pausing and deleting, is this normal behavior Matt?

[mflorell]

post the results of 'screen -r' on the Linux CLI.

Post your keepalives in the astguiclient.conf file.

[riftdawg]

There are several suitable screens on:
2686.asterisk (Detached)
15351.ASTVDadapt (Detached)
15342.ASTlisten (Detached)
15336.ASTupdate (Detached)
2857.ASTfastlog (Detached)
15345.ASTVDauto (Detached)
15339.ASTsend (Detached)
15348.ASTVDremote (Detached)


astgui.conf

VARactive_keepalives => 1234568

[riftdawg]

So I rebooted and now it is working...

[riftdawg]

Ok, I thought I was in the clear, but I was wrong. Dialing out now, but I am getting a Dead AGI error and it is never passing it off to the 8320 extension so I don't see the playback(mymessage). This is what I see in the cli...

-- Executing AGI("Local/my.phone.number@default-1988,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Macro("Local/my.phone.number@default-1988,2", "trunkdial|my.phone.number") in new stack
-- Executing Dial("Local/my.phone.number@default-1988,2", "SIP/trin02/my.phone.number||Tor") in new stack
-- Called trin02/my.phone.number
-- SIP/trin02-091ff7f0 is making progress passing it to Local/my.phone.number@default-1988,2
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/trin02-091ff7f0 answered Local/my.phone.number@default-1988,2
> Channel Local/my.phone.number@default-1988,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("Local/my.phone.number@default-1988,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AGI("Local/my.phone.number@default-1988,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/my.phone.number@default-1988,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing AGI("Local/my.phone.number@default-1988,1", "agi-VDADtransfer.agi|8365") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADtransfer.agi
-- AGI Script agi-VDADtransfer.agi completed, returning 0
-- Executing Hangup("Local/my.phone.number@default-1988,1", "") in new stack
== Spawn extension (default, 8365, 5) exited non-zero on 'Local/my.phone.number@default-1988,1'
-- Executing DeadAGI("Local/my.phone.number@default-1988,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'Local/my.phone.number@default-1988,2' in macro 'trunkdial'
== Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'Local/my.phone.number@default-1988,2'

I used to see it kick over and forward to the ext i specified in remote agent section (which is 8320 in my case). And it would playback the message I told it to.

Matt any ideas? All help is greatly appreciated.

[mflorell]

Have you tried using the sip-silence exten before the transfer AGIs?

There are examples at the bottom of the extensions.conf.sample included with the release.

[riftdawg]

Actually that did it, I thought I had it but i didnt thanks!