Queue problem - goes to agent's voicemail

Not sure I know what’s going on, but for whatever reason, when a person is put into a queue, it drops out and says ‘The person at 2 0 4 is unavailable’. It doesn’t ring the agent or any other agents. The queue has 3 available agents in it on Polycom phones.

Here’s the asterisk full log output:

[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- Executing [900@queues:4] Queue("DAHDI/1-1", "900||||14") in new stack
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- Started music on hold, class 'default', on DAHDI/1-1
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- outgoing agentcall, to agent '203', on 'Local/203@default-20eb,1'
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [203@default:1] Macro("Local/203@default-20eb,2", "stdexten|203|SIP/203") in new stack
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s@macro-stdexten:1] Set("Local/203@default-20eb,2", "__DYNAMIC_FEATURES=") in new stack
[Oct  2 12:38:38] DEBUG[10389] app_macro.c: Executed application: Set
[Oct  2 12:38:38] WARNING[10389] ast_expr2.fl: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
 = 1
 ^
[Oct  2 12:38:38] WARNING[10389] ast_expr2.fl: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s@macro-stdexten:2] GotoIf("Local/203@default-20eb,2", "?5:3") in new stack
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Goto (macro-stdexten,s,3)
[Oct  2 12:38:38] DEBUG[10389] app_macro.c: Executed application: GotoIf
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s@macro-stdexten:3] Dial("Local/203@default-20eb,2", "SIP/203|20|") in new stack
[Oct  2 12:38:38] WARNING[10389] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct  2 12:38:38] VERBOSE[10389] logger.c:   == Everyone is busy/congested at this time (1:0/0/1)
[Oct  2 12:38:38] DEBUG[10389] app_macro.c: Executed application: Dial
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s@macro-stdexten:4] Goto("Local/203@default-20eb,2", "s-CHANUNAVAIL|1") in new stack
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
[Oct  2 12:38:38] DEBUG[10389] app_macro.c: Executed application: Goto
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto("Local/203@default-20eb,2", "s-NOANSWER|1") in new stack
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Goto (macro-stdexten,s-NOANSWER,1)
[Oct  2 12:38:38] DEBUG[10389] app_macro.c: Executed application: Goto
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("Local/203@default-20eb,2", "203|u") in new stack
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- outgoing agentcall, to agent '204', on 'Local/204@default-3270,1'
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [204@default:1] Macro("Local/204@default-3270,2", "stdexten|204|SIP/204") in new stack
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s@macro-stdexten:1] Set("Local/204@default-3270,2", "__DYNAMIC_FEATURES=") in new stack
[Oct  2 12:38:38] DEBUG[10390] app_macro.c: Executed application: Set
[Oct  2 12:38:38] WARNING[10390] ast_expr2.fl: ast_yyerror():  syntax error: syntax error, unexpected '=', expecting $end; Input:
 = 1
 ^
[Oct  2 12:38:38] WARNING[10390] ast_expr2.fl: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s@macro-stdexten:2] GotoIf("Local/204@default-3270,2", "?5:3") in new stack
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Goto (macro-stdexten,s,3)
[Oct  2 12:38:38] DEBUG[10390] app_macro.c: Executed application: GotoIf
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s@macro-stdexten:3] Dial("Local/204@default-3270,2", "SIP/204|20|") in new stack
[Oct  2 12:38:38] WARNING[10390] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct  2 12:38:38] VERBOSE[10390] logger.c:   == Everyone is busy/congested at this time (1:0/0/1)
[Oct  2 12:38:38] DEBUG[10390] app_macro.c: Executed application: Dial
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s@macro-stdexten:4] Goto("Local/204@default-3270,2", "s-CHANUNAVAIL|1") in new stack
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
[Oct  2 12:38:38] DEBUG[10390] app_macro.c: Executed application: Goto
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto("Local/204@default-3270,2", "s-NOANSWER|1") in new stack
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Goto (macro-stdexten,s-NOANSWER,1)
[Oct  2 12:38:38] DEBUG[10390] app_macro.c: Executed application: Goto
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("Local/204@default-3270,2", "204|u") in new stack
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- outgoing agentcall, to agent '206', on 'Local/206@default-956f,1'
[Oct  2 12:38:38] VERBOSE[10391] logger.c:     -- Executing [206@default:1] Dial("Local/206@default-956f,2", "SIP/206") in new stack
[Oct  2 12:38:38] WARNING[10391] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct  2 12:38:38] VERBOSE[10391] logger.c:   == Everyone is busy/congested at this time (1:0/0/1)
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- Agent/204 answered DAHDI/1-1
[Oct  2 12:38:38] DEBUG[10388] chan_agent.c: Hungup, howlong is 0, autologoff is 0
[Oct  2 12:38:38] DEBUG[10388] chan_agent.c: Hungup, howlong is 0, autologoff is 0
[Oct  2 12:38:38] VERBOSE[10388] logger.c:     -- Stopped music on hold on DAHDI/1-1
[Oct  2 12:38:38] VERBOSE[10389] logger.c:     -- <Local/203@default-20eb,2> Playing '/var/spool/asterisk/voicemail/default/203/greet' (language 'en')
[Oct  2 12:38:38] VERBOSE[10389] logger.c:   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Local/203@default-20eb,2' in macro 'stdexten'
[Oct  2 12:38:38] VERBOSE[10389] logger.c:   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Local/203@default-20eb,2'
[Oct  2 12:38:38] VERBOSE[10390] logger.c:     -- <Local/204@default-3270,2> Playing 'vm-theperson' (language 'en')
[Oct  2 12:38:40] VERBOSE[10390] logger.c:     -- <Local/204@default-3270,2> Playing 'digits/2' (language 'en')
[Oct  2 12:38:41] VERBOSE[10390] logger.c:     -- <Local/204@default-3270,2> Playing 'digits/0' (language 'en')
[Oct  2 12:38:41] VERBOSE[10390] logger.c:     -- <Local/204@default-3270,2> Playing 'digits/4' (language 'en')
[Oct  2 12:38:42] VERBOSE[10390] logger.c:     -- <Local/204@default-3270,2> Playing 'vm-isunavail' (language 'en')
[Oct  2 12:38:42] VERBOSE[10390] logger.c:   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Local/204@default-3270,2' in macro 'stdexten'
[Oct  2 12:38:42] VERBOSE[10390] logger.c:   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'Local/204@default-3270,2'
[Oct  2 12:38:42] DEBUG[10388] chan_agent.c: Hungup, howlong is 0, autologoff is 0
[Oct  2 12:38:42] VERBOSE[10388] logger.c:   == Spawn extension (queues, 900, 4) exited non-zero on 'DAHDI/1-1'
[Oct  2 12:38:42] VERBOSE[10388] logger.c:     -- Hungup 'DAHDI/1-1'

extensions.conf

[queues]
exten = 900,1,Set(TIMEOUT(digit)=1)
exten = 900,2,Set(TIMEOUT(response)=1)
exten = 900,3,Background(ZG_Support_Intro)
exten = 900,4,Queue(900||||14)
exten = 900,5,Background(ZG_Support_Busy)
exten = 900,6,Goto(4)

[CallingRule_default]
exten = _1NXXNXXXXXX,1,Macro(trunkdial-failover-0.3,${trunk_2}/+${EXTEN:0},${trunk_1}/${EXTEN:0},trunk_2,trunk_1)

All incoming calls hit that rule. I have a feeling it’s setting it to SIP instead of Local, but I can’t be sure. This seems to happen on and off.

Here are the globals in extensions.conf:

[globals]
CONSOLE => Console/dsp
trunk_1 = Dahdi/g1
trunk_2 = SIP/trunk_2
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =

Should trunk_2 be Local/trunk_2 instead of SIP? Out going calls seem to work fine.

[Oct 2 12:38:38] VERBOSE[10391] logger.c: -- Executing [206@default:1] Dial("Local/206@default-956f,2", "SIP/206") in new stack [Oct 2 12:38:38] WARNING[10391] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)

What is in your sip.conf?

Also:

[Oct 2 12:38:38] WARNING[10390] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 1 ^

[quote=“david55”][Oct 2 12:38:38] VERBOSE[10391] logger.c: -- Executing [206@default:1] Dial("Local/206@default-956f,2", "SIP/206") in new stack [Oct 2 12:38:38] WARNING[10391] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)

What is in your sip.conf?

Also:

[Oct 2 12:38:38] WARNING[10390] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 1 ^ [/quote]

Not too sure how to fix those errors, that’s why I posted. Here’s the sip.conf:

;!
;! Automatically generated configuration file
;! Filename: sip.conf (/etc/asterisk/sip.conf)
;! Generator: Manager
;! Creation Date: Thu Aug 28 10:21:56 2008
;!
[general]
;context = DLPN_DialPlan1
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = no
localnet = 192.168.0.0/255.255.0.0
nat = yes
allowexternaldomains = yes
allowexternalinvites = yes
allowguest = yes
allowsubscribe = no
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
;canreinvite = 
;checkmwi = 
compactheaders = no
;defaultexpiry = 
;domain = 
;dtmfmode = 
dumphistory = no
;externhost = 
;externrefresh = 
;fromdomain = 
g726nonstandard = no
jbenable = no
jbforce = no
;jbimpl = 
jblog = no
;jbmaxsize = 
;jbresyncthreshold = 
;language = 
maxcallbitrate = 384
maxexpiry = 3600
minexpiry = 60
;mohinterpret = 
;mohsuggest = 
;notifymimetype = 
notifyringing = no
pedantic = no
;progressinband = 
promiscredir = no
;realm = 
recordhistory = no
;register = 
;registerattempts = 
;registertimeout = 
relaxdtmf = no
;rtpholdtimeout = 
;rtptimeout = 
sendrpid = no
sipdebug = no
;subscribecontext = 
t1min = 100
t38pt_udptl = no
;tos_audio = 
;tos_sip = 
;tos_video = 
trustrpid = no
;useragent = 
usereqphone = no
videosupport = no
disallow = all
allow = ulaw,gsm,alaw,ilbc

[authentication]

[bandwidth.com_inbound]
host=******
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
reinvite=no
canreinvite=no
context=frombandwidth
nat=yes

You have no [206] section.

That’s in users.conf

Did you get this issue fixed ?

The issue fixed itself. I installed Asterisk Now 1.5 and it’s no longer an issue. My first fix was to redo all the users and queues and that fixed it, also, I made sure the phones weren’t set to DND.

yeah it really look like and old version of macro in the extension.conf … I’ll try to delete the queue and recreate it… anyway thx for your help!

asterisk 1.5 is a stable release ?

If the issue was on/off and you were using persistentmembers in your queues then it might have been that your members were being marked as invalid after a reboot and then when you called them with queue() it just skipped over them all and went to voicemail.

Take a look at this error in your debug:

[Oct 2 12:38:38] WARNING[10389] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1

-bk

Yeah… I know asterisk told us that there’s a syntax error… but where ? in a mcro in extension.conf ?

here’s my error…

I’ve read the file but it didn’t help me a lot… But in my case everything look pretty fine, calls works timerule too even the queue … But I just don’t want to get problem in the future because I didn’t fix this when I saw it :stuck_out_tongue:

I’m not sure if this is a problem with my asterisk config, or with the gui…

Thanks

When is this error occurring? Are you running * with plenty of debug (-cvvvvvv)?

Find out where the warning is being thrown in your dialplan and then paste that part of the dialplan. Please don’t post huge chunks of it cos i’ll never be bothered to read it. Also, if you’re using .ael don’t forget the aelparse command.