If DAHDI is not really configured or chan_dahdi isn't loaded the the error mesage would be "can not create channel of type DAHDI" but here its not the case. Dadhi module is indeed loaded but the DAHDI device is not working properly.<br>
<br><div class="gmail_quote">On Thu, Oct 6, 2011 at 8:49 PM, Gohar Ahmed <span dir="ltr"><<a href="mailto:gohar.ahmed@vopium.com">gohar.ahmed@vopium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Hey,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">How’ve you configured your Outbound trunk ? DAHDI/1/04712527270 : What do you’ve in your dahdi configuration file ! I doubt this “/1” is the culprit or else your DAHDI channel is not really working at all.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Regards,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Gohar A.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"><u></u> <u></u></span></p><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in"><p class="MsoNormal"><b><span style="font-size:10.0pt">From:</span></b><span style="font-size:10.0pt"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>michael k<br>
<b>Sent:</b> Thursday, October 06, 2011 8:46 PM</span></p><div class="im"><br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] PSTN connectivity<u></u><u></u></div>
<p></p></div><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal" style="margin-bottom:12.0pt">Hi All,</p><div><div></div><div class="h5"><br><br> I got a busy message like "all lines are currently busy and please try again later" in call to ZAP trunk. Please help me to resolve this issue <br>
<br><br> == Using SIP RTP TOS bits 184<br> == Using SIP RTP CoS mark 5<br> -- Executing [904712527270@from-internal:1] Macro("SIP/157-00000000", "user-callerid,SKIPTTL,") in new stack<br> -- Executing [s@macro-user-callerid:1] Set("SIP/157-00000000", "AMPUSER=157") in new stack<br>
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/157-00000000", "0?report") in new stack<br> -- Executing [s@macro-user-callerid:3] ExecIf("SIP/157-00000000", "1?Set(REALCALLERIDNUM=157)") in new stack<br>
-- Executing [s@macro-user-callerid:4] Set("SIP/157-00000000", "AMPUSER=157") in new stack<br> -- Executing [s@macro-user-callerid:5] Set("SIP/157-00000000", "AMPUSERCIDNAME=Rojar S") in new stack<br>
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/157-00000000", "0?report") in new stack<br> -- Executing [s@macro-user-callerid:7] Set("SIP/157-00000000", "AMPUSERCID=157") in new stack<br>
-- Executing [s@macro-user-callerid:8] Set("SIP/157-00000000", "CALLERID(all)="Rojar S" <157>") in new stack<br> -- Executing [s@macro-user-callerid:9] ExecIf("SIP/157-00000000", "0?Set(CHANNEL(language)=)") in new stack<br>
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/157-00000000", "1?continue") in new stack<br> -- Goto (macro-user-callerid,s,19)<br> -- Executing [s@macro-user-callerid:19] Set("SIP/157-00000000", "CALLERID(number)=157") in new stack<br>
-- Executing [s@macro-user-callerid:20] Set("SIP/157-00000000", "CALLERID(name)=Rojar S") in new stack<br> -- Executing [s@macro-user-callerid:21] NoOp("SIP/157-00000000", "Using CallerID "Rojar S" <157>") in new stack<br>
-- Executing [904712527270@from-internal:2] Set("SIP/157-00000000", "_NODEST=") in new stack<br> -- Executing [904712527270@from-internal:3] Macro("SIP/157-00000000", "record-enable,157,OUT,") in new stack<br>
-- Executing [s@macro-record-enable:1] GotoIf("SIP/157-00000000", "1?check") in new stack<br> -- Goto (macro-record-enable,s,4)<br> -- Executing [s@macro-record-enable:4] ExecIf("SIP/157-00000000", "0?MacroExit()") in new stack<br>
-- Executing [s@macro-record-enable:5] GotoIf("SIP/157-00000000", "0?Group:OUT") in new stack<br> -- Goto (macro-record-enable,s,15)<br> -- Executing [s@macro-record-enable:15] GotoIf("SIP/157-00000000", "0?IN") in new stack<br>
-- Executing [s@macro-record-enable:16] ExecIf("SIP/157-00000000", "1?MacroExit()") in new stack<br> -- Executing [904712527270@from-internal:4] Macro("SIP/157-00000000", "dialout-trunk,1,04712527270,,") in new stack<br>
-- Executing [s@macro-dialout-trunk:1] Set("SIP/157-00000000", "DIAL_TRUNK=1") in new stack<br> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/157-00000000", "0?sub-pincheck,s,1") in new stack<br>
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/157-00000000", "0?disabletrunk,1") in new stack<br> -- Executing [s@macro-dialout-trunk:4] Set("SIP/157-00000000", "DIAL_NUMBER=04712527270") in new stack<br>
-- Executing [s@macro-dialout-trunk:5] Set("SIP/157-00000000", "DIAL_TRUNK_OPTIONS=tr") in new stack<br> -- Executing [s@macro-dialout-trunk:6] Set("SIP/157-00000000", "OUTBOUND_GROUP=OUT_1") in new stack<br>
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/157-00000000", "0?nomax") in new stack<br> -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/157-00000000", "0?chanfull") in new stack<br>
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/157-00000000", "0?skipoutcid") in new stack<br> -- Executing [s@macro-dialout-trunk:10] Set("SIP/157-00000000", "DIAL_TRUNK_OPTIONS=") in new stack<br>
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/157-00000000", "outbound-callerid,1") in new stack<br> -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/157-00000000", "0?Set(CALLERPRES()=)") in new stack<br>
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/157-00000000", "0?Set(REALCALLERIDNUM=157)") in new stack<br> -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/157-00000000", "1?normcid") in new stack<br>
-- Goto (macro-outbound-callerid,s,6)<br> -- Executing [s@macro-outbound-callerid:6] Set("SIP/157-00000000", "USEROUTCID=") in new stack<br> -- Executing [s@macro-outbound-callerid:7] Set("SIP/157-00000000", "EMERGENCYCID=") in new stack<br>
-- Executing [s@macro-outbound-callerid:8] Set("SIP/157-00000000", "TRUNKOUTCID=") in new stack<br> -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/157-00000000", "1?trunkcid") in new stack<br>
-- Goto (macro-outbound-callerid,s,12)<br> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/157-00000000", "0?Set(CALLERID(all)=)") in new stack<br> -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/157-00000000", "0?Set(CALLERID(all)=)") in new stack<br>
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/157-00000000", "0?Set(CALLERID(all)=)") in new stack<br> -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/157-00000000", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack<br>
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/157-00000000", "0?AGI(fixlocalprefix)") in new stack<br> -- Executing [s@macro-dialout-trunk:13] Set("SIP/157-00000000", "OUTNUM=04712527270") in new stack<br>
-- Executing [s@macro-dialout-trunk:14] Set("SIP/157-00000000", "custom=DAHDI/1") in new stack<br> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/157-00000000", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack<br>
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/157-00000000", "dialout-trunk-predial-hook,") in new stack<br> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/157-00000000", "") in new stack<br>
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/157-00000000", "0?bypass,1") in new stack<br> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/157-00000000", "0?customtrunk") in new stack<br>
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/157-00000000", "DAHDI/1/04712527270,300,") in new stack<br> == Everyone is busy/congested at this time (1:0/0/1)<br> -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/157-00000000", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 66") in new stack<br>
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/157-00000000", "s-CHANUNAVAIL,1") in new stack<br> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)<br> -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/157-00000000", "RC=66") in new stack<br>
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/157-00000000", "66,1") in new stack<br> -- Goto (macro-dialout-trunk,66,1)<br> -- Executing [66@macro-dialout-trunk:1] Goto("SIP/157-00000000", "continue,1") in new stack<br>
-- Goto (macro-dialout-trunk,continue,1)<br> -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/157-00000000", "1?noreport") in new stack<br> -- Goto (macro-dialout-trunk,continue,3)<br>
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/157-00000000", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 - failing through to other trunks") in new stack<br> -- Executing [continue@macro-dialout-trunk:4] Set("SIP/157-00000000", "CALLERID(number)=157") in new stack<br>
-- Executing [904712527270@from-internal:5] Macro("SIP/157-00000000", "outisbusy,") in new stack<br> -- Executing [s@macro-outisbusy:1] Progress("SIP/157-00000000", "") in new stack<br>
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/157-00000000", "0?emergency,1") in new stack<br> -- Executing [s@macro-outisbusy:3] GotoIf("SIP/157-00000000", "0?intracompany,1") in new stack<br>
-- Executing [s@macro-outisbusy:4] Playback("SIP/157-00000000", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack<br> -- <SIP/157-00000000> Playing 'all-circuits-busy-now.gsm' (language 'en')<br>
-- <SIP/157-00000000> Playing 'pls-try-call-later.gsm' (language 'en')<br> -- Executing [s@macro-outisbusy:5] Congestion("SIP/157-00000000", "20") in new stack<br> == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/157-00000000' in macro 'outisbusy'<br>
== Spawn extension (from-internal, 904712527270, 5) exited non-zero on 'SIP/157-00000000'<br> -- Executing [h@from-internal:1] Macro("SIP/157-00000000", "hangupcall") in new stack<br> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/157-00000000", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/157-00000000", "1?skipblkvm") in new stack<br> -- Goto (macro-hangupcall,s,7)<br> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/157-00000000", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br> -- Executing [s@macro-hangupcall:9] Hangup("SIP/157-00000000", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/157-00000000' in macro 'hangupcall'<br>
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/157-00000000'<br><br><br>Michael.k<br><br><br><u></u><u></u></div></div><p></p><div><div></div><div class="h5"><div><p class="MsoNormal">On Fri, Sep 30, 2011 at 10:33 AM, Sam Govind <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>> wrote:<u></u><u></u></p>
<p class="MsoNormal" style="margin-bottom:12.0pt">Hey Warren I thought that these are the complete CLI logs for one call. It started like "<span style="font-size:10.0pt;background:white"> == Using SIP RTP CoS mark 5</span>" and from-internal priority-1 ..So that seemed legit to me. Yeah I too suspect that dialing rules are not being matched and thats why Gotoif's are failing.<u></u><u></u></p>
<div><div><div><p class="MsoNormal">On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby <<a href="mailto:wcselby@selbytech.com" target="_blank">wcselby@selbytech.com</a>> wrote:<u></u><u></u></p></div></div><blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div><div><div><div><p class="MsoNormal"><br>On Thu, Sep 29, 2011 at 7:51 AM, michael k <<a href="mailto:michael@inapp.com" target="_blank">michael@inapp.com</a>> wrote:<u></u><u></u></p><p class="MsoNormal">Thanks for the update. but how do i resolve this issue ? can you help me please ? <u></u><u></u></p>
</div><div><p class="MsoNormal"><br>You didn't provide a full CLI trace of the outgoing call, you only supplied the hangup portion of the call. Please try again.<br><br>Also, what are the dialing rules like in your country? You only have outbound dial patterns setup to handle North American numbers (8+ NXXNXXXXXX or 8+ NXXXXXX).<br>
The Dial Pattern box in the Outbound Rules box is where you define what numbers you want to go out over this trunk. If you dial a number that doesn't match one of these<br>patterns, FreePBX is going to look internally for a dial pattern to match against, and if it doesn't find one there, it will end the call.<br>
<br><br>-- <u></u><u></u></p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt">Thanks,<br>--Warren Selby, dCAP<br><a href="http://www.selbytech.com" target="_blank">http://www.SelbyTech.com</a><br><br><u></u><u></u></p>
</div></div><div><p class="MsoNormal">--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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><u></u><u></u></p></div></blockquote></div><p class="MsoNormal"><br><br>--<br>
_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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><u></u><u></u></p>
</div><p class="MsoNormal"><u></u> <u></u></p></div></div></div></div><br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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></blockquote></div><br>