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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>+1 Bryant – by using the Local/Context you are introducing some overhead to the process, but eliminating the dependence on DAHDI timing (not that there’s anything wrong with that per se, but you can’t control the Space Shuttle with a Bearcat Scanner (or can you?) ).<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Bryant Zimmerman<br><b>Sent:</b> Thursday, November 03, 2011 8:25 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] duration limits in Dial() not being enforced at correct time<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>If you dial to a Local/Context and use your time limits on that and then do your dial to your DAHDI device inside that context does that have any effect on the time limits working. We have used time limits with Local/Context dials and had them work with out any known issues. <o:p></o:p></span></p><div id=divSignature><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks<br><br>Bryant Zimmerman<o:p></o:p></span></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p> </o:p></span></p><div class=MsoNormal align=center style='text-align:center'><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><hr size=2 width="100%" align=center></span></div><p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>: "amit anand" <<a href="mailto:onewaytoconnect@gmail.com">onewaytoconnect@gmail.com</a>><br><b>Sent</b>: Thursday, November 03, 2011 9:18 AM<br><b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br><b>Subject</b>: Re: [asterisk-users] duration limits in Dial() not being enforced at correct time</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><br><br><br><o:p></o:p></span></p><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>On Thu, Nov 3, 2011 at 18:44, Danny Nicholas <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>> wrote:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Please elaborate on your "flavor" of DAHDI and LIBPRI and what type of DAHDI<br>service you are using (PSTN, T1, etc). Speaking from a POTS line point of<br>view, there can easily be a 7-10 second delay in the processing of DAHDI<br>information (which would make your 1347 second call within tolerance).<br><br>-----Original Message-----<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Kingsley Tart<br>Sent: Thursday, November 03, 2011 5:11 AM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: [asterisk-users] duration limits in Dial() not being enforced at<br>correct time<br><br>Hi,<br><br>We're trying to time-limit some calls by specifying L(x:y:z) as an option to<br>the Dial command.<br><br>If we set the limit to a fairly short duration (eg 120 seconds) then<br>Asterisk seems to issue the hangup at about the right time.<br><br>However, for longish calls we're seeing quite a bit of overspill. For<br>example we tried to limit one to 1338 seconds but Asterisk didn't hang up<br>until 1384 seconds after the call was answered.<br><br>Also, the error is not always consistent - a second test call also limited<br>to 1338 seconds was hung up by Asterisk after 1347 seconds.<br><br>We saw this problem with Asterisk 1.6 but we've now tried on Asterisk<br>1.8.6.0 and are having the same problem.<br><br>Here's a log from the Asterisk 1.8.6.0 box for the test call that should<br>have been limited to 1338 seconds but was actually ended after 1384 seconds.<br>The server wasn't carrying any other calls at the time or doing anything<br>else so the load would have been very low.<br><br>[Nov 2 16:47:37] VERBOSE[2029] pbx.c: -- Executing<br>[01476292501@service_nts_v2:57] Dial("DAHDI/i2/7622323283-4",<br>"DAHDI/g1/08451238347,,L(1338000:30000:5000)M(service-nts-v2-register-answer<br>)") in new stack<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > Limit Data for this<br>call:<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > timelimit =<br>1338000 ms (1338.000 s)<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > play_warning = 30000<br>ms (30.000 s)<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > play_to_caller = yes<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > play_to_callee = no<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > warning_freq = 5000<br>ms (5.000 s)<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > start_sound =<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > warning_sound =<br>/var/lib/asterisk/sounds/bespoke/beep_200ms<br>[Nov 2 16:47:37] VERBOSE[2029] features.c: > end_sound =<br>[Nov 2 16:47:37] VERBOSE[2029] sig_pri.c: -- Requested transfer<br>capability: 0x00 - SPEECH<br>[Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- Called<br>DAHDI/g1/08451238347<br>[Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is<br>proceeding passing it to DAHDI/i2/7622323283-4<br>[Nov 2 16:47:37] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3 is<br>ringing<br>[Nov 2 16:47:38] VERBOSE[2029] app_dial.c: -- DAHDI/i1/08451238347-3<br>answered DAHDI/i2/7622323283-4<br>[Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing<br>[s@macro-service-nts-v2-register-answer:1] NoOp("DAHDI/i1/08451238347-3",<br>"ANSWER MACRO") in new stack<br>[Nov 2 16:47:38] VERBOSE[2029] pbx.c: -- Executing<br>[s@macro-service-nts-v2-register-answer:2] AGI("DAHDI/i1/08451238347-3",<br>"agi://<a href="http://127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,uniqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1" target="_blank">127.0.0.1:4573/ServiceNTSV2,mode=answered,uniqueID=1320252457.17_1,un<br>iqueIDB=1320252457.18,ddi=08451238347,Goto=agiOK1</a>") in new stack<br>[Nov 2 16:47:39] VERBOSE[2029] res_agi.c: -- AGI Script Executing<br>Application: (Goto) Options: (agiOK1)<br>[Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto<br>(macro-service-nts-v2-register-answer,s,7)<br>[Nov 2 16:47:39] VERBOSE[2029] res_agi.c: --<br><DAHDI/i1/08451238347-3>AGI Script agi://<a href="http://127.0.0.1:4573/ServiceNTSV2" target="_blank">127.0.0.1:4573/ServiceNTSV2</a><br>completed, returning 0<br>[Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing<br>[s@macro-service-nts-v2-register-answer:7] GotoIf("DAHDI/i1/08451238347-3",<br>"1?agiOK2") in new stack<br>[Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Goto<br>(macro-service-nts-v2-register-answer,s,13)<br>[Nov 2 16:47:39] VERBOSE[2029] pbx.c: -- Executing<br>[s@macro-service-nts-v2-register-answer:13] NoOp("DAHDI/i1/08451238347-3",<br>"register-answer macro finished") in new stack<br>[Nov 2 16:47:39] VERBOSE[2029] chan_dahdi.c: -- Native bridging<br>DAHDI/i2/7622323283-4 and DAHDI/i1/08451238347-3<br>[Nov 2 17:10:42] VERBOSE[2029] pbx.c: -- Executing [h@service_nts_v2:1]<br>NoOp("DAHDI/i2/7622323283-4", "number HANGING UP ...<br>CHANNEL=DAHDI/i2/7622323283-4, channel1=1320252457.17_1, channel2=,<br>HANGUPCAUSE=16, UNIQUEID=1320252457.17") in new stack<br><br>Is this a known problem and are there any workarounds?<br><br>--<br>Cheers,<br>Kingsley.<br><span style='color:#888888'><br><br><span class=hoenzb>--</span><br><span class=hoenzb>_____________________________________________________________________</span><br><span class=hoenzb>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --</span><br><span class=hoenzb>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span class=hoenzb> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a></span><br><br><span class=hoenzb>asterisk-users mailing list</span><br><span class=hoenzb>To UNSUBSCRIBE or update options visit:</span><br><span class=hoenzb> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span><br><br><br><span class=hoenzb>--</span><br><span class=hoenzb>_____________________________________________________________________</span><br><span class=hoenzb>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --</span><br><span class=hoenzb>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span class=hoenzb> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a></span><br><br><span class=hoenzb>asterisk-users mailing list</span><br><span class=hoenzb>To UNSUBSCRIBE or update options visit:</span><br><span class=hoenzb> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></span><o:p></o:p></span></p></div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><br><br><br>Hi you can use Absoulte timeout to set the time limit feature for the channel<br clear=all><o:p></o:p></span></p><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>-- <o:p></o:p></span></p><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Amit Anand<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:13.5pt;font-family:"Arial","sans-serif"'>+91 9818559898</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p></div></body></html>