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lolu<br>
I reformated the output so it was easier to understand. I attached the
word document for you.<br>
on the below line:<br>
<br>
<p><tt><span style="font-family: "Courier New";"><span style=""> </span>--
Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in
new stack<br>
<span style=""> </span>-- Requested transfer capability:
0x00 - SPEECH<o:p></o:p></span></tt></p>
<p><small><span style="font-size: 12pt; font-family: "Courier New";"><tt><span
style=""> </span>-- Called 1/8774957</tt></span></small></p>
<p><span style="font-size: 12pt; font-family: "Courier New";"><tt><span
style=""> </span>-- Zap/1-1 is proceeding passing it
to SIP/7871-f813 <b>Don't know what to do if second ROSE component is
of type 0x6</b></tt></span></p>
<h6><span style="font-size: 12pt; font-family: "Courier New";"></span></h6>
<h6><span style="font-size: 12pt; font-family: "Courier New";"></span></h6>
<span style="font-size: 12pt; font-family: "Courier New";"><!--[if !supportLineBreakNewLine]--><!--[endif]--></span><span
style="font-size: 12pt; font-family: "Courier New";"><span style=""></span><!--[if !supportLineBreakNewLine]--></span><span
style="font-size: 12pt; font-family: "Courier New";"></span><span
style="font-size: 12pt; font-family: "Courier New";"></span><span
style="font-size: 12pt; font-family: "Courier New";"></span>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..<br>
<br>
in any case, the question you must answer is '<b>what is the second
ROSE component</b>'??? and <b>why is of type 0x6</b>???<br>
how is it set and by what component?<br>
hope that moves you closer to the ultimate resolution...<br>
daveC<br>
<br>
<br>
Lolu Gbenga wrote:
<blockquote
cite="mid72ca844c0712200827q3edd47b7m1715fbcd79a9da3f@mail.gmail.com"
type="cite">Hi All<br>
I FOUND OUT THAT THE ATTACHMENT WAS NOT SENT WITH THE MAIL.<br>
FIND BELOW THE OUTPUT USING asterisk -vvvr command for EXTERNAL calls
that gave the ouput ALL TRUNKS ARE BUSY PLEASE TRY YOUR CALL LATER.<br>
<pre><tt>
<tt>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></tt></tt></pre>
<br>
ALSO FIND BELOW THE OUTPUT using asterisk -vvvr command FOR INTERNAL
calls that rang.
<br>
<br>
<pre><tt><tt>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
<a href="http://127.0.0.1">127.0.0.1</a>
-- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager 'admin' logged off from <a href="http://127.0.0.1">127.0.0.1</a>
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.
</tt></tt></pre>
<br>
<br>
<br>
<div class="gmail_quote">On Dec 20, 2007 5:09 PM, Lolu Gbenga <<a
href="mailto:olugbenga1@gmail.com">olugbenga1@gmail.com</a>> wrote:
<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi
all,<br>
I am grateful for our contribution so far .<br>
<br>
I followed dave advise and i have the attached file using the aterisk
-vvvvr when a call is made.
<br>
<br>
I attached two files.<br>
<br>
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.
<br>
<br>
The second attachment is when i made internal calls and the phone rang.<br>
<br>
Please,i will be expecting your replies for further directions.<br>
<br>
Best Regards
<div>
<div class="Wj3C7c"><br>
<br>
<br>
<div class="gmail_quote">On Dec 20, 2007 2:58 PM, Steve Totaro <
<a href="mailto:stotaro@first-notification.com" target="_blank">stotaro@first-notification.com</a>>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">What
is the output of ztconfig from the Linux command line? What does<br>
your zaptel.conf and zapata.conf look like? What is the relevant part<br>
of extensions.conf (the dialout section that fails). Also from the
CLI,
<br>
it would be most helpful to post the output you get when dialing out<br>
fails. I don't think it is a network issue at all, I think your configs<br>
need some work.<br>
<br>
Thanks,<br>
<font color="#888888">Steve Totaro
<br>
</font>
<div><br>
Lolu Gbenga wrote:<br>
> Good Day<br>
><br>
</div>
<div>> Find attached the relevant portions of the asterisk CLI.<br>
><br>
> Please,which portion of the extension .conf should i send ?
<br>
><br>
> It is connected via RJ 45 connector to an E1 modem to the telco
company.<br>
><br>
> I use E1 link.<br>
><br>
> I will appreciate your reply.<br>
><br>
> Best Regards<br>
><br>
><br>
> On Dec 18, 2007 4:02 PM, dave cantera <
<a href="mailto:david.cantera@iacnet.net" target="_blank">david.cantera@iacnet.net</a><br>
</div>
<div>
<div>> <mailto:<a href="mailto:david.cantera@iacnet.net"
target="_blank">david.cantera@iacnet.net</a>> > wrote:
<br>
><br>
> lolu,<br>
> sounds more like a telco/itsp problem then *.<br>
> I would<br>
> tcpdump -i eth0 port 5060<br>
> to make sure it is actually going out... change 5060 if you
have<br>
> changed<br>
> your port to your itsp, of course.<br>
> to see what is going on as well as the other debugging notes
mentioned<br>
> in this thread.<br>
> daveC<br>
><br>
> Lolu Gbenga wrote:
<br>
> > Good Day all<br>
> ><br>
> > Please I am having some issues on my voip asterisk server<br>
> ><br>
> > I make internal calls on extensions configured ie
extension 192 can
<br>
> > call extension 195 etc<br>
> ><br>
> > But each time i try to make calls outside the extension
ie calling a<br>
> > GSM or an external line ,i always hear this response "all
trunk
<br>
> calls<br>
> > are busy please try your call again later"<br>
> ><br>
> > Please how can i resolve this problem .<br>
> ><br>
> > I will appreciate your response.
<br>
> ><br>
> > Best Regards<br>
> ><br>
> > Success<br>
> ><br>
> > _______________________________________________<br>
> > --Bandwidth and Colocation Provided by <a
href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>
> ><br>
> > asterisk-users mailing list<br>
> > To UNSUBSCRIBE or update options visit:<br>
> >
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
> ><br>
> ><br>
> ><br>
> ><br>
><br>
> --<br>
> My wife's sister is in California.<br>
> I should buy her a Videophone2008!<br>
><br>
> Truly, The Next Best Thing to Being There!<br>
> --<br>
><br>
> WorldWideVideoPhones.com
<br>
> 856.380.0894<br>
><br>
><br>
><br>
><br>
><br>
<br>
<br>
_______________________________________________<br>
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</div>
</div>
</blockquote>
</div>
<br>
</div>
</div>
</blockquote>
</div>
<br>
<pre wrap="">
<hr size="4" width="90%">
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<pre wrap="">
<hr size="4" width="90%">
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.503 / Virus Database: 269.17.2 - Release Date: 12/14/2007 12:00 AM
</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="132">--
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
</pre>
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