[asterisk-users] Problem with extensions in IVR and queues

Thu Jul 1 14:11:03 CDT 2010

Hi, we've just been able to find the problem. Apparently it was related to the softphone. We've installed another one and the call is performed ok.
Thanks!

Anahi Ludueña

From: a_luduena at hotmail.com
To: asterisk-users at lists.digium.com
Date: Wed, 30 Jun 2010 19:59:14 +0000
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues

Ups, sorry, that CLI output is related to my other problem (the options of IVR doesn't responde when the call is from landline or cell phone).
I'll put the correct CLI output...
Thanks,

Anahi Ludueña

From: a_luduena at hotmail.com
To: asterisk-users at lists.digium.com
Date: Wed, 30 Jun 2010 19:50:00 +0000
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues

This is the CLI output, the dialplan is the one that the Elastix creates when somebody sets the followme... I don't know what part you want I post here...
Thanks,

    -- Executing [4010 at from-internal:1] GotoIf("SIP/9050-001185aa", "0?ext-local|4010|1") in new stack
    -- Executing [4010 at from-internal:2] Macro("SIP/9050-001185aa", "user-callerid|") in new stack
    -- Executing [s at macro-user-callerid:1] Set("SIP/9050-001185aa", "AMPUSER=9050") in new stack
    -- Executing [s at macro-user-callerid:2] GotoIf("SIP/9050-001185aa", "0?report") in new stack
    -- Executing [s at macro-user-callerid:3] ExecIf("SIP/9050-001185aa", "1|Set|REALCALLERIDNUM=9050") in new stack
    -- Executing [s at macro-user-callerid:4] Set("SIP/9050-001185aa", "AMPUSER=9050") in new stack
    -- Executing [s at macro-user-callerid:5] Set("SIP/9050-001185aa", "AMPUSERCIDNAME=CALLPBX") in new stack
    -- Executing [s at macro-user-callerid:6] GotoIf("SIP/9050-001185aa", "0?report") in new stack
    -- Executing [s at macro-user-callerid:7] Set("SIP/9050-001185aa", "AMPUSERCID=9050") in new stack
    -- Executing [s at macro-user-callerid:8] Set("SIP/9050-001185aa", "CALLERID(all)="CALLPBX" <9050>") in new stack
    -- Executing [s at macro-user-callerid:9] ExecIf("SIP/9050-001185aa", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s at macro-user-callerid:10] GotoIf("SIP/9050-001185aa", "0?continue") in new stack
    -- Executing [s at macro-user-callerid:11] Set("SIP/9050-001185aa", "__TTL=64") in new stack
    -- Executing [s at macro-user-callerid:12] GotoIf("SIP/9050-001185aa", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s at macro-user-callerid:19] NoOp("SIP/9050-001185aa", "Using CallerID "CALLPBX" <9050>") in new stack
    -- Executing [4010 at from-internal:3] GotoIf("SIP/9050-001185aa", "1?skipdb") in new stack
    -- Goto (from-internal,4010,5)
    -- Executing [4010 at from-internal:5] Set("SIP/9050-001185aa", "__NODEST=") in new stack
    -- Executing [4010 at from-internal:6] Set("SIP/9050-001185aa", "__BLKVM_OVERRIDE=BLKVM/4010/SIP/9050-001185aa") in new stack
    -- Executing [4010 at from-internal:7] Set("SIP/9050-001185aa", "__BLKVM_BASE=4010") in new stack
    -- Executing [4010 at from-internal:8] Set("SIP/9050-001185aa", "DB(BLKVM/4010/SIP/9050-001185aa)=TRUE") in new stack
    -- Executing [4010 at from-internal:9] Set("SIP/9050-001185aa", "RRNODEST=") in new stack
    -- Executing [4010 at from-internal:10] Set("SIP/9050-001185aa", "__NODEST=4010") in new stack
    -- Executing [4010 at from-internal:11] Set("SIP/9050-001185aa", "RecordMethod=Group") in new stack
    -- Executing [4010 at from-internal:12] Macro("SIP/9050-001185aa", "record-enable|4010|Group") in new stack
    -- Executing [s at macro-record-enable:1] GotoIf("SIP/9050-001185aa", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s at macro-record-enable:4] AGI("SIP/9050-001185aa", "recordingcheck|20100630-154030|1277926830.37214") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s at macro-record-enable:5] MacroExit("SIP/9050-001185aa", "") in new stack
    -- Executing [4010 at from-internal:13] Set("SIP/9050-001185aa", "RingGroupMethod=ringallv2") in new stack
    -- Executing [4010 at from-internal:14] Set("SIP/9050-001185aa", "_FMGRP=4010") in new stack
    -- Executing [4010 at from-internal:15] GotoIf("SIP/9050-001185aa", "0?doconfirm") in new stack
    -- Executing [4010 at from-internal:16] Macro("SIP/9050-001185aa", "dial|20|tr|4010") in new stack
    -- Executing [s at macro-dial:1] GotoIf("SIP/9050-001185aa", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s at macro-dial:3] AGI("SIP/9050-001185aa", "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 'CALLPBX' number is '9050'
  dialparties.agi: USE_CONFIRMATION:  'FALSE'
  dialparties.agi: RINGGROUP_INDEX:   ''
  dialparties.agi: Methodology of ring is  'ringallv2'
    --  dialparties.agi: Added extension 4010 to extension map
       >  dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 20
       >  dialparties.agi: fmgrp_totalprering: 22
       >  dialparties.agi: found extension in pre-ring and array
       >  dialparties.agi: ringallv2 ring times: REALPRERING: 22, PRERING: 2
    --  dialparties.agi: Extension 4010 cf is disabled
    --  dialparties.agi: Extension 4010 do not disturb is disabled
       >  dialparties.agi: extnum 4010 has:  cw: 0; hascfb: 0 [] hascfu: 0 []
  dialparties.agi: ExtensionState: 4
  dialparties.agi: Extension 4010 has ExtensionState: 4
    --  dialparties.agi: Checking CW and CFB status for extension 4010
    --  dialparties.agi: dbset CALLTRACE/4010 to 9050
    --  dialparties.agi: Filtered ARG3: 4010
       >  dialparties.agi: NODEST: 4010 adding M(auto-blkvm) to dialopts: trM(auto-blkvm)
       >  dialparties.agi: NODEST: 4010 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 at macro-dial:7] Dial("SIP/9050-001185aa", "SIP/4010|22|trM(auto-blkvm)") in new stack
Really destroying SIP dialog '1544c4ea374acd44596154e42c84825b at 127.0.0.1' Method: INVITE
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s at macro-dial:8] Set("SIP/9050-001185aa", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s at macro-dial:9] GosubIf("SIP/9050-001185aa", "0?CHANUNAVAIL|1") in new stack
    -- Executing [4010 at from-internal:17] Goto("SIP/9050-001185aa", "nextstep") in new stack
    -- Goto (from-internal,4010,19)
    -- Executing [4010 at from-internal:19] Set("SIP/9050-001185aa", "RingGroupMethod=") in new stack
    -- Executing [4010 at from-internal:20] GotoIf("SIP/9050-001185aa", "0?nodest") in new stack
    -- Executing [4010 at from-internal:21] Set("SIP/9050-001185aa", "__NODEST=") in new stack
    -- Executing [4010 at from-internal:22] DBdel("SIP/9050-001185aa", "BLKVM/4010/SIP/9050-001185aa") in new stack
    -- DBdel: family=BLKVM, key=4010/SIP/9050-001185aa
    -- Executing [4010 at from-internal:23] Goto("SIP/9050-001185aa", "ivr-3|s|1") in new stack
    -- Goto (ivr-3,s,1)
    -- Executing [s at ivr-3:1] Set("SIP/9050-001185aa", "MSG=custom/CALL-English") in new stack
    -- Executing [s at ivr-3:2] Set("SIP/9050-001185aa", "LOOPCOUNT=0") in new stack
    -- Executing [s at ivr-3:3] Set("SIP/9050-001185aa", "__DIR-CONTEXT=default") in new stack
    -- Executing [s at ivr-3:4] Set("SIP/9050-001185aa", "_IVR_CONTEXT_ivr-3=") in new stack
    -- Executing [s at ivr-3:5] Set("SIP/9050-001185aa", "_IVR_CONTEXT=ivr-3") in new stack
    -- Executing [s at ivr-3:6] GotoIf("SIP/9050-001185aa", "0?begin") in new stack
    -- Executing [s at ivr-3:7] Answer("SIP/9050-001185aa", "") in new stack
    -- Executing [s at ivr-3:8] Wait("SIP/9050-001185aa", "1") in new stack
    -- Executing [s at ivr-3:9] Set("SIP/9050-001185aa", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing [s at ivr-3:10] Set("SIP/9050-001185aa", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing [s at ivr-3:11] Set("SIP/9050-001185aa", "__IVR_RETVM=") in new stack
    -- Executing [s at ivr-3:12] ExecIf("SIP/9050-001185aa", "1|Background|custom/CALL-English") in new stack
    -- <SIP/9050-001185aa> Playing 'custom/CALL-English' (language 'en')
Really destroying SIP dialog '48d34342645adfa70265fa8e5291c266 at XXX.XXX.XXX.XXX' Method: OPTIONS
Really destroying SIP dialog '200bf37a463ff4bb5673ba4720cec6c1 at XXX.XXX.XXX.XXX' Method: OPTIONS
Really destroying SIP dialog '24d9c31a44a206f216d2c142338fbf36 at XXX.XXX.XXX.XXX' Method: NOTIFY
    -- Got SIP response 603 "Declined (no dialog)" back from YYY.YYY.YYY.YYY
Really destroying SIP dialog '42debf4b37838b98708590dc6e42548c at XXX.XXX.XXX.XXX' Method: NOTIFY
    -- Executing [s at ivr-3:13] WaitExten("SIP/9050-001185aa", "|") in new stack
  == Spawn extension (ivr-3, s, 13) exited non-zero on 'SIP/9050-001185aa'
    -- Executing [h at ivr-3:1] Hangup("SIP/9050-001185aa", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9050-001185aa'

Anahi Ludueña

From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Wed, 30 Jun 2010 14:08:19 -0500
Subject: Re: [asterisk-users] Problem with extensions in IVR and queues

Can you post the dialplan section and CLI
output from one of these calls?

From:
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Wednesday, June 30, 2010
2:05 PM

To:
asterisk-users at lists.digium.com

Subject: Re: [asterisk-users]
Problem with extensions in IVR and queues

Thanks Danny, but I don't know
what I should do to fix it...

Could you help me?

Anahi
Ludueña

From: danny at debsinc.com

To: asterisk-users at lists.digium.com

Date: Wed, 30 Jun 2010 10:33:31 -0500

Subject: Re: [asterisk-users] Problem with extensions in IVR and queues

Sounds like you are getting a “dial
without bridge” – asterisk dials x and make the connection, but because the
bridge doesn’t happen for what ever reason, the call disconnects like no one
ever answered.

From:
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anahi Ludueña

Sent: Wednesday, June 30, 2010
10:29 AM

To:
asterisk-users at lists.digium.com

Subject: [asterisk-users] Problem
with extensions in IVR and queues

Hi people, 

we have some extensions which are included in the IVRs and/or queues.
Everything works fine, but the calls done from these extensions are hang up
after 30 o 35 seconds. If they are not included in the IVR or queues, the calls
are performed well.

Do you know if there is something else to set?

Thanks,

Anahi
Ludueña

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