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<br> -- Executing Macro("SIP/7871-f813", "dialout-trunk|1|018774957||")<br> in new sta ck<br> -- Executing GotoIf("SIP/7871-f813", "1?3:2") in new stack
<br> -- Goto (macro-dialout-trunk,s,3)<br> -- Executing Macro("SIP/7871-f813", "user-callerid") in new stack<br> -- Executing Set("SIP/7871-f813", "AMPUSER=7871") in new stack
<br> -- Executing Set("SIP/7871-f813", "EMERGENCYCID=7871") in new stack<br> -- Executing Set("SIP/7871-f813", "AMPUSERCIDNAME=7871") in new<br> stack<br> -- Executing GotoIf("SIP/7871-f813", "0?6") in new stack
<br> -- Executing Set("SIP/7871-f813", "CALLERID(all)="7871" <7871>") in<br> new stack<br> -- Executing NoOp("SIP/7871-f813", "Using CallerID "7871" <7871>")
<br> in new stack<br> -- Executing Macro("SIP/7871-f813", "record-enable|7871|OUT") in<br> new stack<br> -- Executing GotoIf("SIP/7871-f813", "0 > 0?2:4") in new stack<br> -- Goto (macro-record-enable,s,4)
<br> -- Executing AGI("SIP/7871-f813",<br> "recordingcheck|20051006-001624|1128554184. 8") in new stack<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
<br> recordingcheck|20051006-001624|1128554184.8: Outbound recording not<br> enabled<br> -- AGI Script recordingcheck completed, returning 0<br> -- Executing NoOp("SIP/7871-f813", "No recording needed") in new
<br> stack<br> -- Executing Macro("SIP/7871-f813", "outbound-callerid|1") in new<br> stack<br> -- Executing Set("SIP/7871-f813", "USEROUTCID=7871") in new stack<br> -- Executing GotoIf("SIP/7871-f813", "1?4") in new stack
<br> -- Goto (macro-outbound-callerid,s,4)<br> -- Executing GotoIf("SIP/7871-f813", "0?6") in new stack<br> -- Executing Set("SIP/7871-f813", "CALLERID(all)=7871") in new<br>
stack<br> -- Executing GotoIf("SIP/7871-f813", "1?8") in new stack<br> -- Goto (macro-outbound-callerid,s,8)<br> -- Executing NoOp("SIP/7871-f813", "CallerID set to "" <7871>") in
<br> new stack<br> -- Executing Set("SIP/7871-f813", "GROUP()=OUT_1") in new stack<br> -- Executing GotoIf("SIP/7871-f813", "0?108") in new stack<br> -- Executing Set("SIP/7871-f813", "DIAL_NUMBER=018774957") in new
<br> stack<br> -- Executing Set("SIP/7871-f813", "DIAL_TRUNK=1") in new stack<br> -- Executing AGI("SIP/7871-f813", "fixlocalprefix") in new stack<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
<br> fixlocalprefix: Removed prefix. New number: 8774957<br> -- AGI Script fixlocalprefix completed, returning 0<br> -- Executing Set("SIP/7871-f813", "OUTNUM=8774957") in new stack<br> -- Executing Set("SIP/7871-f813", "custom=ZAP/1") in new stack
<br> -- Executing GotoIf("SIP/7871-f813", "0?16") in new stack<br> -- Executing Dial("SIP/7871-f813", "ZAP/1/8774957|120|W") in new<br> stack<br> -- Requested transfer capability: 0x00 - SPEECH
<br> -- Called 1/8774957<br> -- Zap/1-1 is proceeding passing it to SIP/7871-f813<br>Don't know what to do if second ROSE component is of type 0x6<br> -- Channel 0/1, span 1 got hangup request<br> -- Hungup 'Zap/1-1'
<br> == Everyone is busy/congested at this time (1:0/0/1)<br> -- Executing Goto("SIP/7871-f813", "s-CHANUNAVAIL|1") in new stack<br> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)<br> -- Executing NoOp("SIP/7871-f813", "Dial failed due to
<br> CHANUNAVAIL") in new s tack<br> -- Executing Macro("SIP/7871-f813", "outisbusy|") in new stack<br> -- Executing Playback("SIP/7871-f813", "all-circuits-busy-now") in
<br> new stack<br> -- Playing 'all-circuits-busy-now' (language 'en')<br> -- Executing Playback("SIP/7871-f813", "pls-try-call-later") in new<br> stack<br> -- Playing 'pls-try-call-later' (language 'en')
<br> -- Executing Macro("SIP/7871-f813", "hangupcall") in new stack<br> -- Executing ResetCDR("SIP/7871-f813", "w") in new stack<br> -- Executing NoCDR("SIP/7871-f813", "") in new stack
<br> -- Executing Wait("SIP/7871-f813", "5") in new stack<br> -- Executing Hangup("SIP/7871-f813", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
<br> 'SIP/7871-f813' in macro<br> 'hangupcall'<br> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on<br> 'SIP/7871-f813' in macro
<br> 'outisbusy'<br> == Spawn extension (macro-hangupcall, s, 4) exited non-zero on<br> 'SIP/7871-f813'<br>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<br> -- Executing Macro("SIP/7871-bb64", "exten-vm|novm|7874") in new<br> stack<br> -- Executing Macro("SIP/7871-bb64", "user-callerid") in new stack
<br> -- Executing Set("SIP/7871-bb64", "AMPUSER=7871") in new stack<br> -- Executing Set("SIP/7871-bb64", "EMERGENCYCID=7871") in new stack<br> -- Executing Set("SIP/7871-bb64", "AMPUSERCIDNAME=7871") in new
<br> stack<br> -- Executing GotoIf("SIP/7871-bb64", "0?6") in new stack<br> -- Executing Set("SIP/7871-bb64", "CALLERID(all)="7871" <7871>") in<br> new stack<br>
-- Executing NoOp("SIP/7871-bb64", "Using CallerID "7871" <7871>")<br> in new stack<br> -- Executing Set("SIP/7871-bb64", "FROMCONTEXT=exten-vm") in new<br> stack
<br> -- Executing Macro("SIP/7871-bb64", "record-enable|7874|IN") in new<br> stack<br> -- Executing GotoIf("SIP/7871-bb64", "0 > 0?2:4") in new stack<br> -- Goto (macro-record-enable,s,4)
<br> -- Executing AGI("SIP/7871-bb64",<br> "recordingcheck|20051006-002614|1128554774. 10") in new<br> stack<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
<br> recordingcheck|20051006-002614|1128554774.10: Inbound recording not<br> enabled<br> -- AGI Script recordingcheck completed, returning 0<br> -- Executing NoOp("SIP/7871-bb64", "No recording needed") in new
<br> stack<br> -- Executing Macro("SIP/7871-bb64", "dial|15|tr|7874") in new stack<br> -- Executing AGI("SIP/7871-bb64", "dialparties.agi") in new stack<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
<br> -- dialparties.agi: priority = 1<br> -- dialparties.agi: callingani2 = 0<br> -- dialparties.agi: accountcode =<br> -- dialparties.agi: channel = SIP/7871-bb64<br> -- dialparties.agi: callerid = 7871
<br> -- dialparties.agi: context = macro-dial<br> -- dialparties.agi: callington = 0<br> -- dialparties.agi: dnid = 7874<br> -- dialparties.agi: request = dialparties.agi<br> -- dialparties.agi: calleridname = 7871
<br> -- dialparties.agi: extension = s<br> -- dialparties.agi: language = en<br> -- dialparties.agi: uniqueid = 1128554774.10<br> -- dialparties.agi: callingpres = 0<br> -- dialparties.agi: type = SIP<br>
-- dialparties.agi: rdnis = unknown<br> -- dialparties.agi: callingtns = 0<br> -- dialparties.agi: enhanced = 0.0<br> dialparties.agi: Caller ID name is '7871' number is '7871'<br> dialparties.agi
: Methodology of ring is 'none'<br> -- dialparties.agi: Added extension 7874 to extension map<br> -- dialparties.agi: Extension 7874 cf is disabled<br> -- dialparties.agi: Extension 7874 do not disturb is disabled
<br> -- dialparties.agi: Checking CW and CFB status for extension 7874<br> == Parsing '/etc/asterisk/manager.conf': Found<br> == Parsing '/etc/asterisk/manager_custom.conf': Found<br> == Manager 'admin' logged on from
<a href="http://127.0.0.1">127.0.0.1</a><br> -- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS<br> == Manager 'admin' logged off from <a href="http://127.0.0.1">127.0.0.1</a><br> dialparties.agi: Extension 7874 is available...skipping checks
<br> -- dialparties.agi: DbSet CALLTRACE/7874 to 7871<br> -- AGI Script dialparties.agi completed, returning 0<br> -- Executing Dial("SIP/7871-bb64", "SIP/7874|15|tr") in new stack<br> -- Called 7874
<br> -- SIP/7874-5b48 is ringing<br> == Spawn extension (macro-dial, s, 10) exited non-zero on<br> 'SIP/7871-bb64' in ma cro 'dial'<br> == Spawn extension (macro-dial, s, 10) exited non-zero on
<br> 'SIP/7871-bb64' in ma cro 'exten-vm'<br> == Spawn extension (macro-dial, s, 10) exited non-zero on<br> 'SIP/7871-bb64'<br>asterisk1*CLI><br><br>
<br>THANKS SO MUCH I WILL BE EXPECTING YOUR REPLY.<br></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></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></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
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