[asterisk-users] All trunk are busy please try your call again later
Lolu Gbenga
olugbenga1 at gmail.com
Thu Dec 20 10:27:23 CST 2007
Hi All
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls that
gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.
Verbosity is at least 3
-- Executing Macro("SIP/7871-f813", "dialout-trunk|1|018774957||")
in new sta ck
-- Executing GotoIf("SIP/7871-f813", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/7871-f813", "user-callerid") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSER=7871") in new stack
-- Executing Set("SIP/7871-f813", "EMERGENCYCID=7871") in new stack
-- Executing Set("SIP/7871-f813", "AMPUSERCIDNAME=7871") in new
stack
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
-- Executing Set("SIP/7871-f813", "CALLERID(all)="7871" <7871>") in
new stack
-- Executing NoOp("SIP/7871-f813", "Using CallerID "7871" <7871>")
in new stack
-- Executing Macro("SIP/7871-f813", "record-enable|7871|OUT") in
new stack
-- Executing GotoIf("SIP/7871-f813", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/7871-f813",
"recordingcheck|20051006-001624|1128554184.
8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20051006-001624|1128554184.8: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/7871-f813", "No recording needed") in new
stack
-- Executing Macro("SIP/7871-f813", "outbound-callerid|1") in new
stack
-- Executing Set("SIP/7871-f813", "USEROUTCID=7871") in new stack
-- Executing GotoIf("SIP/7871-f813", "1?4") in new stack
-- Goto (macro-outbound-callerid,s,4)
-- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
-- Executing Set("SIP/7871-f813", "CALLERID(all)=7871") in new
stack
-- Executing GotoIf("SIP/7871-f813", "1?8") in new stack
-- Goto (macro-outbound-callerid,s,8)
-- Executing NoOp("SIP/7871-f813", "CallerID set to "" <7871>") in
new stack
-- Executing Set("SIP/7871-f813", "GROUP()=OUT_1") in new stack
-- Executing GotoIf("SIP/7871-f813", "0?108") in new stack
-- Executing Set("SIP/7871-f813", "DIAL_NUMBER=018774957") in new
stack
-- Executing Set("SIP/7871-f813", "DIAL_TRUNK=1") in new stack
-- Executing AGI("SIP/7871-f813", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Removed prefix. New number: 8774957
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("SIP/7871-f813", "OUTNUM=8774957") in new stack
-- Executing Set("SIP/7871-f813", "custom=ZAP/1") in new stack
-- Executing GotoIf("SIP/7871-f813", "0?16") in new stack
-- Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 1/8774957
-- Zap/1-1 is proceeding passing it to SIP/7871-f813
Don't know what to do if second ROSE component is of type 0x6
-- Channel 0/1, span 1 got hangup request
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Goto("SIP/7871-f813", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/7871-f813", "Dial failed due to
CHANUNAVAIL") in new s tack
-- Executing Macro("SIP/7871-f813", "outisbusy|") in new stack
-- Executing Playback("SIP/7871-f813", "all-circuits-busy-now") in
new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/7871-f813", "pls-try-call-later") in new
stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing Macro("SIP/7871-f813", "hangupcall") in new stack
-- Executing ResetCDR("SIP/7871-f813", "w") in new stack
-- Executing NoCDR("SIP/7871-f813", "") in new stack
-- Executing Wait("SIP/7871-f813", "5") in new stack
-- Executing Hangup("SIP/7871-f813", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/7871-f813' in macro
'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/7871-f813' in macro
'outisbusy'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/7871-f813'
asterisk1*CLI>
ALSO FIND BELOW THE OUTPUT using asterisk -vvvr command FOR INTERNAL calls
that rang.
Verbosity is at least 3
-- Executing Macro("SIP/7871-bb64", "exten-vm|novm|7874") in new
stack
-- Executing Macro("SIP/7871-bb64", "user-callerid") in new stack
-- Executing Set("SIP/7871-bb64", "AMPUSER=7871") in new stack
-- Executing Set("SIP/7871-bb64", "EMERGENCYCID=7871") in new stack
-- Executing Set("SIP/7871-bb64", "AMPUSERCIDNAME=7871") in new
stack
-- Executing GotoIf("SIP/7871-bb64", "0?6") in new stack
-- Executing Set("SIP/7871-bb64", "CALLERID(all)="7871" <7871>") in
new stack
-- Executing NoOp("SIP/7871-bb64", "Using CallerID "7871" <7871>")
in new stack
-- Executing Set("SIP/7871-bb64", "FROMCONTEXT=exten-vm") in new
stack
-- Executing Macro("SIP/7871-bb64", "record-enable|7874|IN") in new
stack
-- Executing GotoIf("SIP/7871-bb64", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/7871-bb64",
"recordingcheck|20051006-002614|1128554774.
10") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20051006-002614|1128554774.10: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/7871-bb64", "No recording needed") in new
stack
-- Executing Macro("SIP/7871-bb64", "dial|15|tr|7874") in new stack
-- Executing AGI("SIP/7871-bb64", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: priority = 1
-- dialparties.agi: callingani2 = 0
-- dialparties.agi: accountcode =
-- dialparties.agi: channel = SIP/7871-bb64
-- dialparties.agi: callerid = 7871
-- dialparties.agi: context = macro-dial
-- dialparties.agi: callington = 0
-- dialparties.agi: dnid = 7874
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: calleridname = 7871
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: uniqueid = 1128554774.10
-- dialparties.agi: callingpres = 0
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: callingtns = 0
-- dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name is '7871' number is '7871'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 7874 to extension map
-- dialparties.agi: Extension 7874 cf is disabled
-- dialparties.agi: Extension 7874 do not disturb is disabled
-- dialparties.agi: Checking CW and CFB status for extension 7874
== 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 7874 is available...skipping checks
-- dialparties.agi: DbSet CALLTRACE/7874 to 7871
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/7871-bb64", "SIP/7874|15|tr") in new stack
-- Called 7874
-- SIP/7874-5b48 is ringing
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/7871-bb64' in ma cro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/7871-bb64' in ma cro
'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/7871-bb64'
asterisk1*CLI>
THANKS SO MUCH I WILL BE EXPECTING YOUR REPLY.
On Dec 20, 2007 5:09 PM, Lolu Gbenga <olugbenga1 at gmail.com> wrote:
> Hi all,
> I am grateful for our contribution so far .
>
> I followed dave advise and i have the attached file using the aterisk
> -vvvvr when a call is made.
>
> I attached two files.
>
> One of the attached file is for the external call,which replied with the
> PROBLEM all trunks are busy now,please try your call again later.
>
> The second attachment is when i made internal calls and the phone rang.
>
> Please,i will be expecting your replies for further directions.
>
> Best Regards
>
>
>
> On Dec 20, 2007 2:58 PM, Steve Totaro < stotaro at first-notification.com>
> wrote:
>
> > What is the output of ztconfig from the Linux command line? What does
> > your zaptel.conf and zapata.conf look like? What is the relevant part
> > of extensions.conf (the dialout section that fails). Also from the CLI,
> >
> > it would be most helpful to post the output you get when dialing out
> > fails. I don't think it is a network issue at all, I think your configs
> > need some work.
> >
> > Thanks,
> > Steve Totaro
> >
> > Lolu Gbenga wrote:
> > > Good Day
> > >
> > > Find attached the relevant portions of the asterisk CLI.
> > >
> > > Please,which portion of the extension .conf should i send ?
> > >
> > > It is connected via RJ 45 connector to an E1 modem to the telco
> > company.
> > >
> > > I use E1 link.
> > >
> > > I will appreciate your reply.
> > >
> > > Best Regards
> > >
> > >
> > > On Dec 18, 2007 4:02 PM, dave cantera < david.cantera at iacnet.net
> > > <mailto:david.cantera at iacnet.net> > wrote:
> > >
> > > lolu,
> > > sounds more like a telco/itsp problem then *.
> > > I would
> > > tcpdump -i eth0 port 5060
> > > to make sure it is actually going out... change 5060 if you have
> > > changed
> > > your port to your itsp, of course.
> > > to see what is going on as well as the other debugging notes
> > mentioned
> > > in this thread.
> > > daveC
> > >
> > > Lolu Gbenga wrote:
> > > > Good Day all
> > > >
> > > > Please I am having some issues on my voip asterisk server
> > > >
> > > > I make internal calls on extensions configured ie extension 192
> > can
> > > > call extension 195 etc
> > > >
> > > > But each time i try to make calls outside the extension ie
> > calling a
> > > > GSM or an external line ,i always hear this response "all trunk
> > > calls
> > > > are busy please try your call again later"
> > > >
> > > > Please how can i resolve this problem .
> > > >
> > > > I will appreciate your response.
> > > >
> > > > Best Regards
> > > >
> > > > Success
> > > >
> > > > _______________________________________________
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> > > >
> > > >
> > > >
> > > >
> > >
> > > --
> > > My wife's sister is in California.
> > > I should buy her a Videophone2008!
> > >
> > > Truly, The Next Best Thing to Being There!
> > > --
> > >
> > > WorldWideVideoPhones.com
> > > 856.380.0894
> > >
> > >
> > >
> > >
> > >
> >
> >
> > _______________________________________________
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> >
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