lolu
I reformated the output so it was easier to understand. I attached the
word document for you.
on the below line:
--
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
it looks
like this is where it determines it can't proceed... also, there are
many tests along the way... we don't know about the
questions/conditions and if that effects it or not... probably not..
in any case, the question you must answer is 'what is the second
ROSE component'??? and why is of type 0x6???
how is it set and by what component?
hope that moves you closer to the ultimate resolution...
daveC
Lolu Gbenga wrote:
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 < [EMAIL PROTECTED]> 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 <
[EMAIL PROTECTED]>
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 <
[EMAIL PROTECTED]
> <mailto: [EMAIL PROTECTED]> > 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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--
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
|
