Calls have silence when coming through Queues Asterisk 1.4.19

QueueMetrics > General Asterisk configuration

(1/2) > >>

rneubauer:
I am getting this in the asterisk log....

VERBOSE[17835] logger.c: -- DBdel: Error deleting key from database.
[Jun 4 11:49:12] DEBUG[17835] app_macro.c: Executed application: DBDel
[Jun 4 11:49:12] DEBUG[17835] app_dial.c: Macro exited with status 0
[Jun 4 11:49:12] DEBUG[18424] app_macro.c: Executed application: DBDel
[Jun 4 11:49:12] DEBUG[18424] app_dial.c: Macro exited with status 0
[Jun 4 11:49:12] DEBUG[18423] app_queue.c: Dunno what to do with control type -1
[Jun 4 11:49:12] VERBOSE[18423] logger.c: -- Agent/206 answered SIP/205-092bcf18
[Jun 4 11:49:12] VERBOSE[18423] logger.c: -- Stopped music on hold on SIP/205-092bcf18
[Jun 4 11:49:12] WARNING[18423] app_queue.c: The device state of this queue member, Agent/206, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings

Help!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!

Rick

rneubauer:
Here is is smoe additional info I found in the CLI....

new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("Local/206@from-internal-adb6,2", "TTL: 64 ARG1: 206") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("Local/206@from-internal-adb6,2", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("Local/206@from-internal-adb6,2", "__TTL=63") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("Local/206@from-internal-adb6,2", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("Local/206@from-internal-adb6,2", "Using CallerID "Sales:Colleen Hunt" <205>") in new stack
-- Executing [s@macro-exten-vm:2] Set("Local/206@from-internal-adb6,2", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("Local/206@from-internal-adb6,2", "VMBOX=206") in new stack
-- Executing [s@macro-exten-vm:4] Set("Local/206@from-internal-adb6,2", "EXTTOCALL=206") in new stack
-- Executing [s@macro-exten-vm:5] Set("Local/206@from-internal-adb6,2", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("Local/206@from-internal-adb6,2", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("Local/206@from-internal-adb6,2", "RT=20") in new stack
-- Executing [s@macro-exten-vm:8] Macro("Local/206@from-internal-adb6,2", "record-enable|206|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("Local/206@from-internal-adb6,2", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("Local/206@from-internal-adb6,2", "recordingcheck|20080604-114708|1212598028.399") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080604-114708|1212598028.399: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("Local/206@from-internal-adb6,2", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("Local/206@from-internal-adb6,2", "dial|20|tr|206") in new stack
-- Executing [s@macro-dial:1] GotoIf("Local/206@from-internal-adb6,2", "0?dial") in new stack
-- Executing [s@macro-dial:2] SetMusicOnHold("Local/206@from-internal-adb6,2", "default") in new stack
-- Executing [s@macro-dial:3] AGI("Local/206@from-internal-adb6,2", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Sales:Colleen Hunt' number is '205'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 206 to extension map
-- dialparties.agi: Extension 206 cf is disabled
-- dialparties.agi: Extension 206 do not disturb is disabled
> dialparties.agi: extnum 206 has: cw: 1; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
-- dialparties.agi: dbset CALLTRACE/206 to 205
-- dialparties.agi: Filtered ARG3: 206
> dialparties.agi: NODEST: 100 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
> dialparties.agi: NODEST: 100 blkvm enabled macro already in dialopts: trM(auto-blkvm)
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("Local/206@from-internal-adb6,2", "SIP/206|20|trM(auto-blkvm)") in new stack
-- Called 206
-- Agent/206 is ringing
-- SIP/206-092bcf18 is ringing
-- SIP/206-092bcf18 answered Local/206@from-internal-adb6,2
-- Executing [s@macro-auto-blkvm:1] Set("SIP/206-092bcf18", "__MACRO_RESULT=") in new stack
-- Executing [s@macro-auto-blkvm:2] Set("SIP/206-092bcf18", "__CWIGNORE=") in new stack
-- Executing [s@macro-auto-blkvm:3] DBdel("SIP/206-092bcf18", "BLKVM/100/SIP/205-09312a28") in new stack
-- DBdel: family=BLKVM, key=100/SIP/205-09312a28
-- Agent/206 answered SIP/205-09312a28
-- Stopped music on hold on SIP/205-09312a28
-- User hung up
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/6309323200-092eb368' in macro 'vm'
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/6309323200-092eb368' in macro 'exten-vm'
== Spawn extension (macro-vm, s-BUSY, 3) exited non-zero on 'SIP/6309323200-092eb368'
connect*CLI>

QueueMetrics:

--- Quote from: rneubauer on June 04, 2008, 18:58:14 ---I am getting this in the asterisk log....

[Jun 4 11:49:12] WARNING[18423] app_queue.c: The device state of this queue member, Agent/206, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings

--- End quote ---

I quote this from the UPGRADE.txt file in Asterisk 1.4.8:

* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.

Example:

[general]
limitonpeer = yes

[peername]
type=friend
call-limit=10

Is this your case?

rneubauer:
This particular box is TB 2.6.7, and running 1.4.19.

I did ADD the limitonpeer = yes on the sip_nat.conf file, then all the agents have call limit=50.

Am I missing something?

rneubauer:
i figured out a few more things....

If I setup a direct route into the the Queue. Then I can pick it up perfectly.. But if I have a DID go to an IVR, and have it setup let's say as "6" to go to the IVR, then the call is just dead air...but when I hang up the phone call, I get justa split second of audo from call....

So I took a look at the queues then selves, and here is what I found:

Event When Called: When this option is set to YES, the following manager events will be generated: AgentCalled, AgentDump, AgentConnect and AgentComplete.

set this to yes to have it report the correct events to the Queuelog for Queuemetrics...

Member Status:When if this is option is set to YES, the following manager event will be generated: QueueMemberStatus

Set this to yes as well...

Skip Busy Agents:When set to Yes, agents who are on an occupied phone will be skipped as if the line were returning busy. This means that Call Waiting or multi-line phones will not be presented with the call and in the various hunt style ring strategies, the next agent will be attempted.

I set this to yes so people only got one call at a time...

that got it working!!!!!! Yay me!!!!!

Rick

Navigation

[0] Message Index

[#] Next page

Go to full version