[Asterisk-Users] FW: Re: Delay on Phone ringing

Mark Hulber asterisk-admin at hulber.com
Tue Feb 28 06:00:27 MST 2006


The only time I see recorded in your log is that of the recording check

    -- Executing AGI("Zap/1-1", "recordingcheck|20060227-131600|1141046151.2") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled


which doesn't seem to take any time. Only you would know at what phase 
the dialplan was in at each point of the 12 seconds. How long did it 
take before this took place:

    -- Starting simple switch on 'Zap/1-1'

How long did this phase take:

    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    -- AGI Script dialparties.agi completed, returning 0


MARK.

Ash Thakrar wrote:
>
> Hi,
>
> I have just joined this mail list yesterday and have been searching 
> the Asterisk wiki prior to posting this question.
>
> Unfortunately I am not sure if I am searching at the correct places, 
> so I do apologise if this has been posted before.
>
> I have currently been tasked to roll out VoIP phones through out our 
> office as the current proprietary Panasonic PBX has no more channels.
>
> Thus I have installed Asterisk at home on VIA SP13000,512Mb Ram and using 
> 2 x Digum TDM400P cards with both having 4x TDM40B FXO modules.
>
> I have rolled out 12 x Snom320 phones & 1 x Snom360 in the office.
>
> For the test phase, I wanted to use the current PBX, Therefore Port 1 
> of the TDM is currently connected to one of the POTS extensions which 
> is spare on the current PBX.
>
> Current problems I am facing in the test phase:
>
> Whenever I call from outside e.g. from the fax line (separate line) or 
> my mobile, to the main number setup on the Trunk, I get a delay of 
> around 12sec before the VoIP phone actually rings, although the phones 
> connected to the current PBX, ring immediately.
>
> I have attached the output file and noticed that the DBget is trying 
> to find ‘something’ in the AstDB, would that be causing the delay?
>
> Or am I looking at the wrong place altogether.
>
> Please Help
>
> Regards
>
> Ash Thakrar
>
> ------------------------------------------------------------------------
>
> asterisk1*CLI> soft hangup Zap/1-1
> Requested Hangup on channel 'Zap/1-1'
>   == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
>   == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
>     -- Hungup 'Zap/1-1'
>     -- Starting simple switch on 'Zap/1-1'
>     -- Executing GotoIf("Zap/1-1", "1?from-pstn-reghours|s|1:") in new stack
>     -- Goto (from-pstn-reghours,s,1)
>     -- Executing GotoIf("Zap/1-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack
>     -- Goto (from-pstn-reghours,s,2)
>     -- Executing Answer("Zap/1-1", "") in new stack
>     -- Executing Wait("Zap/1-1", "1") in new stack
>     -- Executing SetVar("Zap/1-1", "intype=EXT-220") in new stack
>     -- Executing Cut("Zap/1-1", "intype=intype|-|1") in new stack
>     -- Executing GotoIf("Zap/1-1", "1?7:9") in new stack
>     -- Goto (from-pstn-reghours,s,7)
>     -- Executing Wait("Zap/1-1", "3") in new stack
>     -- Executing Goto("Zap/1-1", "ext-local|220|1") in new stack
>     -- Goto (ext-local,220,1)
>     -- Executing Macro("Zap/1-1", "exten-vm|novm|220") in new stack
>     -- Executing Macro("Zap/1-1", "user-callerid") in new stack
>     -- Executing DBget("Zap/1-1", "AMPUSER=DEVICE//user") in new stack
>     -- DBget: varname=AMPUSER, family=DEVICE, key=/user
>     -- DBget: Value not found in database.
>     -- Executing DBget("Zap/1-1", "AMPUSERCIDNAME=AMPUSER//cidname") in new stack
>     -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname
>     -- DBget: Value not found in database.
>     -- Executing GotoIf("Zap/1-1", "1?5") in new stack
>     -- Goto (macro-user-callerid,s,5)
>     -- Executing NoOp("Zap/1-1", "Using CallerID ") in new stack
>     -- Executing SetVar("Zap/1-1", "FROMCONTEXT=exten-vm") in new stack
>     -- Executing Macro("Zap/1-1", "record-enable|220|IN") in new stack
>     -- Executing GotoIf("Zap/1-1", "0 > 0?2:4") in new stack
>     -- Goto (macro-record-enable,s,4)
>     -- Executing AGI("Zap/1-1", "recordingcheck|20060227-131600|1141046151.2") in new stack
>     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
>   recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled
>     -- AGI Script recordingcheck completed, returning 0
>     -- Executing NoOp("Zap/1-1", "No recording needed") in new stack
>     -- Executing Macro("Zap/1-1", "dial|15|tr|220") in new stack
>     -- Executing GotoIf("Zap/1-1", "0?4:2") in new stack
>     -- Goto (macro-dial,s,2)
>     -- Executing GotoIf("Zap/1-1", "0?5:4") in new stack
>     -- Goto (macro-dial,s,4)
>     -- Executing AGI("Zap/1-1", "dialparties.agi") in new stack
>     -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
>     --  dialparties.agi: priority = 4
>     --  dialparties.agi: callingani2 = 0
>     --  dialparties.agi: accountcode =
>     --  dialparties.agi: channel = Zap/1-1
>     --  dialparties.agi: callerid = unknown
>     --  dialparties.agi: context = macro-dial
>     --  dialparties.agi: callington = 0
>     --  dialparties.agi: dnid = unknown
>     --  dialparties.agi: request = dialparties.agi
>     --  dialparties.agi: calleridname = unknown
>     --  dialparties.agi: extension = s
>     --  dialparties.agi: language = en
>     --  dialparties.agi: uniqueid = 1141046151.2
>     --  dialparties.agi: callingpres = 0
>     --  dialparties.agi: type = Zap
>     --  dialparties.agi: rdnis = unknown
>     --  dialparties.agi: callingtns = 0
>     --  dialparties.agi: enhanced = 0.0
>   dialparties.agi: Caller ID is not set
>   dialparties.agi: Methodology of ring is  'none'
>     --  dialparties.agi: Added extension 220 to extension map
>     --  dialparties.agi: Extension 220 cf is disabled
>     --  dialparties.agi: Extension 220 do not disturb is disabled
>     --  dialparties.agi: Checking CW and CFB status for extension 220
>   == Parsing '/etc/asterisk/manager.conf': Found
>   == Parsing '/etc/asterisk/manager_custom.conf': Found
>   == Manager 'admin' logged on from 127.0.0.1
>     --  dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
>   == Manager 'admin' logged off from 127.0.0.1
>   dialparties.agi: Extension 220 is available...skipping checks
>     --  dialparties.agi: DbDel CALLTRACE/220 - Caller ID is not defined
>     -- AGI Script dialparties.agi completed, returning 0
>     -- Executing Dial("Zap/1-1", "SIP/220|15|tr") in new stack
>     -- Called 220
>     -- SIP/220-1f31 is ringing
>     -- SIP/220-1f31 is ringing
>     -- SIP/220-1f31 is ringing
>     -- SIP/220-1f31 is ringing
>     -- SIP/220-1f31 is ringing
>     -- SIP/220-1f31 is ringing
>     -- Nobody picked up in 15000 ms
>     -- Executing GotoIf("Zap/1-1", "0?s-NOANSWER|1") in new stack
>     -- Executing GotoIf("Zap/1-1", "1?s-NOANSWER|1") in new stack
>     -- Goto (macro-exten-vm,s-NOANSWER,1)
>     -- Executing Congestion("Zap/1-1", "") in new stack
> asterisk1*CLI> soft hangup Zap/1-1
> Requested Hangup on channel 'Zap/1-1'
>   == Spawn extension (macro-exten-vm, s-NOANSWER, 1) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
>   == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
>     -- Hungup 'Zap/1-1'
>   
> ------------------------------------------------------------------------
>
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