"No one is in your session: 8600051" On Just One Dialer

[johnt]

6 Servers (1 DB/Web, 5 Dialers)
VERSION: 2.8-440a
BUILD: 140521-2101


I installed two new dialers a while back using VICIBox 5.0.3. One is working great, the other not so much...

Agent logs in, receives the call from the system and answers. Within 15 seconds or so the Agent GUI reports "No one is in your session: 8600051" However, the connection is still fine and will remain fine for as long as the agent remains logged in, ie not until the agent hits "Go Back" and then "Logout" will the connection hang up. All the other dialers are working fine.

The one thing different about this server is that is it the voicemail server. I just mention this, because I am trying to find anything different about the two...

Here is snippet from Asterisk -
[Oct 10 06:18:22] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 10 06:18:28] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 10 06:18:28] == Using SIP RTP TOS bits 184
[Oct 10 06:18:28] == Using SIP RTP CoS mark 5
[Oct 10 06:18:33] > Channel SIP/cc04108-00000002 was answered.
[Oct 10 06:18:33] -- Executing [8600052@default:1] MeetMe("SIP/cc04108-00000002", "8600052,F") in new stack
[Oct 10 06:18:33] == Parsing '/etc/asterisk/meetme.conf': [Oct 10 06:18:33] == Found
[Oct 10 06:18:33] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Oct 10 06:18:33] == Found
[Oct 10 06:18:33] -- Created MeetMe conference 1023 for conference '8600052'
[Oct 10 06:18:33] -- <SIP/cc04108-00000002> Playing 'conf-onlyperson.slin' (language 'en')
[Oct 10 06:18:34] == Manager 'sendcron' logged off from 127.0.0.1

Nothing else appears in the CLI until the agent logs out... I have poked around in /var/log/astguiclient

The one thing I see "funny" is in the action_full log, notice how the Channel on the call to the agent shows the correct SIP channel, but when the agent logs out, the channel is "empty" It appears on other servers that are working that the Channel on the Hang Up action is always populated...

Here is a snippet from
2014-10-10 6:18:28|62688708|0|1|
Action: Originate
Channel: SIP/cc04108
Context: default
Exten: 8600052
Priority: 1
Callerid: "S1410100618278600052" <8009821589>

|
2014-10-10 6:18:34|62688708|1|||
2014-10-10 6:18:34|62688708|2|||
2014-10-10 6:25:20|62688709|0|1|
Action: Hangup
Channel: <----- see nothing here!!!


I have compared the routing tables on the "bad" dialer with a good dialer and they appear to be exactly the same. And all of the equipment sits on the same subnet...

I would prefer to not have to wipe this server and start over, so any ideas are greatly appreciated. Thanks!!!

[boybawang]

what is the output of screen -ls

[johnt]

I think I just figured it out. I am about to test it in production, but the Asterisk version for the server in question had a space in front of it... Never would have noticed it on the front end, but it sticks out in the database...