<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'>
Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ?<br><br>campbx1*CLI> module show like timing<br>Module Description Use Count <br>res_timing_pthread.so pthread Timing Interface 0 <br>res_timing_timerfd.so Timerfd Timing Interface 1 <br>res_timing_dahdi.so DAHDI Timing Interface 0 <br>3 modules loaded<br><br><br><hr id="stopSpelling">From: nic@njcolledge.net<br>To: asterisk-users@lists.digium.com<br>Date: Fri, 13 May 2011 15:11:19 +0000<br>Subject: Re: [asterisk-users] 1.8 and prematuremedia problem<br><br>
<meta http-equiv="Content-Type" content="text/html; charset=unicode">
<meta name="Generator" content="Microsoft SafeHTML">
<style>
.ExternalClass p.ecxMsoNormal, .ExternalClass li.ecxMsoNormal, .ExternalClass div.ecxMsoNormal
{margin-bottom:.0001pt;font-size:12.0pt;font-family:'Times New Roman','serif';}
.ExternalClass a:link, .ExternalClass span.ecxMsoHyperlink
{color:blue;text-decoration:underline;}
.ExternalClass a:visited, .ExternalClass span.ecxMsoHyperlinkFollowed
{color:purple;text-decoration:underline;}
.ExternalClass p
{margin-right:0cm;margin-left:0cm;font-size:12.0pt;font-family:'Times New Roman','serif';}
.ExternalClass span.ecxEmailStyle18
{font-family:'Calibri','sans-serif';color:#1F497D;letter-spacing:0pt;}
.ExternalClass .ecxMsoChpDefault
{font-size:10.0pt;}
@page WordSection1
{size:612.0pt 792.0pt;}
.ExternalClass div.ecxWordSection1
{page:WordSection1;}
</style>
<div class="ecxWordSection1">
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);">At the asterisk CLI type “module show like timing”</span></p>
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p>
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);">Whichever has a use-count >1 is the one you are using.</span></p>
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p>
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);">Nic.</span></p>
<p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p>
<div>
<div style="border-right: medium none; border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0cm 0cm;">
<p class="ecxMsoNormal"><b><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';" lang="EN-US">From:</span></b><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';" lang="EN-US"> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b>On Behalf Of </b>satish patel<br>
<b>Sent:</b> 13 May 2011 16:03<br>
<b>To:</b> tbskyd@gmail.com; asterisk-users<br>
<b>Subject:</b> Re: [asterisk-users] 1.8 and prematuremedia problem</span></p>
</div>
</div>
<p class="ecxMsoNormal"> </p>
<p class="ecxMsoNormal"><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';">Thanks for reply,<br>
<br>
How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ?<br>
<br>
-S<br>
<br>
> Date: Fri, 13 May 2011 22:13:47 +0800<br>
> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem<br>
> From: tbskyd@gmail.com<br>
> To: satish_lx@hotmail.com; asterisk-users@lists.digium.com<br>
> <br>
> hi:<br>
> I am using 64bit scientific linux 6 with default kernel. my<br>
> loading is quite low, maybe 1~10 concurrent calls. I remember last<br>
> time I have unstable problem about timer.<br>
> my linux now use HPET clock. and asterisk use res_timing_dahdi instead<br>
> of the default res_timing_timerfd. I don't know if these are related<br>
> to you problem. hope you can find the key point to make a stable<br>
> asterisk.<br>
> <br>
> Regards,<br>
> tbskyd<br>
> <br>
> 2011/5/13 Satish Patel <satish_lx@hotmail.com>:<br>
> > Glad you solved it. Now I'm having high CPU load issue. I don't know why but<br>
> > sometime my asterisk process reached ~150% CPU load and just locked no calls<br>
> > nothing only solution is kill -9<br>
> ><br>
> > I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because<br>
> > of low through put ?? Which OS are you using?<br>
> ><br>
> > --<br>
> > Sent from my iPhone<br>
> ><br>
> > On May 12, 2011, at 9:31 PM, d tbsky <tbskyd@gmail.com> wrote:<br>
> ><br>
> >> hi:<br>
> >> sorry. the issue number is 19268. not 19628.<br>
> >> sorry about that!!<br>
> >><br>
> >> Regards,<br>
> >> tbskyd<br>
> >><br>
> >> 2011/5/13 d tbsky <tbskyd@gmail.com>:<br>
> >>><br>
> >>> hi:<br>
> >>> I report my issue as issue 19628.<br>
> >>> it is fixed and I run asterisk 1.8 in production now.<br>
> >>> thanks a lot for your help!<br>
> >>><br>
> >>> Regards,<br>
> >>> tbskyd<br>
> >>><br>
> >>> 2011/5/11 d tbsky <tbskyd@gmail.com>:<br>
> >>>><br>
> >>>> hi:<br>
> >>>> ok I will create a bug report. and I found I still need<br>
> >>>> "prematuremedia=no" in asterisk 1.6.2.18.<br>
> >>>> yesterday I was testing at home with zoiper softphone + iax. today I<br>
> >>>> test snom hardware sip phone and found that "prematuremedia=no" is<br>
> >>>> still necessary.<br>
> >>>><br>
> >>>> Regards,<br>
> >>>> tbskyd<br>
> >>>><br>
> >>>><br>
> >>>> 2011/5/11 satish patel <satish_lx@hotmail.com>:<br>
> >>>>><br>
> >>>>> I am sorry about that but its interesting it doesn't work with 1.8 SVN<br>
> >>>>><br>
> >>>>> I would say please report this bug so that way you can track issue, And<br>
> >>>>> may<br>
> >>>>> be in future it help us :)<br>
> >>>>><br>
> >>>>> -S<br>
> >>>>><br>
> >>>>>> Date: Wed, 11 May 2011 01:31:34 +0800<br>
> >>>>>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem<br>
> >>>>>> From: tbskyd@gmail.com<br>
> >>>>>> To: asterisk-users@lists.digium.com; satish_lx@hotmail.com<br>
> >>>>>><br>
> >>>>>> hi:<br>
> >>>>>> that issue is marked as fixed, so no more comment can be added :(<br>
> >>>>>> anyway, I try the following combination:<br>
> >>>>>> 1.8.3.2 + sig_pri patch<br>
> >>>>>> 1.8 svn which already has sig_pri patched<br>
> >>>>>> 1.8.4 + libpri patch (another unofficial patch in issue 18868)<br>
> >>>>>><br>
> >>>>>> but none works.<br>
> >>>>>><br>
> >>>>>> finally I downgrade to 1.6.2.18 and I found everything works. I don't<br>
> >>>>>> even need to set "prematuremedia" with 1.6.2.18.<br>
> >>>>>> so I think I will need to stay with 1.6.2 a little longer...<br>
> >>>>>><br>
> >>>>>> thanks a lot for your help!!<br>
> >>>>>><br>
> >>>>>> Regards,<br>
> >>>>>> tbskyd<br>
> >>>>>><br>
> >>>>>> 2011/5/10 satish patel <satish_lx@hotmail.com>:<br>
> >>>>>>><br>
> >>>>>>> Also i would say add comment on following issue if after patch you<br>
> >>>>>>> having<br>
> >>>>>>> issue, That way it help community to fine tune patch.<br>
> >>>>>>><br>
> >>>>>>> https://issues.asterisk.org/view.php?id=18868<br>
> >>>>>>><br>
> >>>>>>> Good luck<br>
> >>>>>>><br>
> >>>>>>><br>
> >>>>>>>> From: satish_lx@hotmail.com<br>
> >>>>>>>> To: tbskyd@gmail.com<br>
> >>>>>>>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem<br>
> >>>>>>>> Date: Tue, 10 May 2011 07:43:47 -0400<br>
> >>>>>>>> CC: asterisk-users@lists.digium.com<br>
> >>>>>>>><br>
> >>>>>>>> I have applied this patch in 1.8 svn branch and it works great for<br>
> >>>>>>>> me.<br>
> >>>>>>>><br>
> >>>>>>>> I have nothing special configuration just simple dial command for<br>
> >>>>>>>> outgoing call.<br>
> >>>>>>>><br>
> >>>>>>>> Also check there are progress=yes option in chan_dahdi<br>
> >>>>>>>><br>
> >>>>>>>> --<br>
> >>>>>>>> Sent from my iPhone<br>
> >>>>>>>><br>
> >>>>>>>> On May 10, 2011, at 5:58 AM, d tbsky <tbskyd@gmail.com> wrote:<br>
> >>>>>>>><br>
> >>>>>>>>> hi:<br>
> >>>>>>>>> I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not<br>
> >>>>>>>>> apply to 1.8.3.2 or 1.8.4-rc3).<br>
> >>>>>>>>> but the situation is the same. do I need to play with other options<br>
> >>>>>>>>> with the patch? or I need<br>
> >>>>>>>>> newer asterisk versions to solve the problem?<br>
> >>>>>>>>> thanks a lot for information!!<br>
> >>>>>>>>><br>
> >>>>>>>>> 2011/5/10 d tbsky <tbskyd@gmail.com>:<br>
> >>>>>>>>>><br>
> >>>>>>>>>> hi:<br>
> >>>>>>>>>> thanks a lot for your quick reply. I saw that patch and think that<br>
> >>>>>>>>>> it was already included in 1.8.3.<br>
> >>>>>>>>>> now I know it will be included in 1.8.5.<br>
> >>>>>>>>>> I will try it and thanks again for your kindly help!!<br>
> >>>>>>>>>><br>
> >>>>>>>>>> 2011/5/10 Satish Patel <satish_lx@hotmail.com>:<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> Apply this patch https://issues.asterisk.org/view.php?id=18868<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> --<br>
> >>>>>>>>>>> Sent from my iPhone<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> On May 9, 2011, at 9:57 PM, d tbsky <tbskyd@gmail.com> wrote:<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>>> hi:<br>
> >>>>>>>>>>>> our current connection is below:<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> sip phone<--->asterisk<---->alcatel PBX<---->PSTN<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> asterisk and alcatel PBX is connected via E1 isdn-pri.<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> when I use sip phone to dial outside PSTN world:<br>
> >>>>>>>>>>>> 1. with 1.4 it is fine.<br>
> >>>>>>>>>>>> 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or<br>
> >>>>>>>>>>>> sip<br>
> >>>>>>>>>>>> phone can not hear the ring and the beginning of the PSTN voice.<br>
> >>>>>>>>>>>> 3. with 1.8.3.2, I can not hear ring and the beginning of the<br>
> >>>>>>>>>>>> PSTN<br>
> >>>>>>>>>>>> voice. I try to play options with "prematuremedia" and<br>
> >>>>>>>>>>>> "progressinband". but I can not find working settings.<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> I don't know what other options I can try.<br>
> >>>>>>>>>>>> thank a lot for information!!<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> --<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> _____________________________________________________________________<br>
> ><br>
> ><br>
> >>>>>>>><br>
> >>>>>>>><br>
> >>>>>>>>>>>> -- Bandwidth and Colocation Provided by http://www.api-<br>
> >>>>>>>>>>>> digital.com --<br>
> >>>>>>>>>>>> New to Asterisk? Join us for a live introductory webinar every<br>
> >>>>>>>>>>>> Thurs:<br>
> >>>>>>>>>>>> http://www.asterisk.org/hello<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>>> asterisk-users mailing list<br>
> >>>>>>>>>>>> To UNSUBSCRIBE or update options visit:<br>
> >>>>>>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>
> >>>>>>>>>>>><br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> --<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> _____________________________________________________________________<br>
> ><br>
> ><br>
> >>>>>>>><br>
> >>>>>>>><br>
> >>>>>>>>>>> -- Bandwidth and Colocation Provided by<br>
> >>>>>>>>>>> http://www.api-digital.com<br>
> >>>>>>>>>>> --<br>
> >>>>>>>>>>> New to Asterisk? Join us for a live introductory webinar every<br>
> >>>>>>>>>>> Thurs:<br>
> >>>>>>>>>>> http://www.asterisk.org/hello<br>
> >>>>>>>>>>><br>
> >>>>>>>>>>> asterisk-users mailing list<br>
> >>>>>>>>>>> To UNSUBSCRIBE or update options visit:<br>
> >>>>>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>
> >>>>>>>>>>><br>
> >>>>>>>>>><br>
> >>>>>>>>><br>
> >>>>>>><br>
> >>>>>>> --<br>
> >>>>>>> _____________________________________________________________________<br>
> ><br>
> ><br>
> >>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>
> >>>>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> >>>>>>> http://www.asterisk.org/hello<br>
> >>>>>>><br>
> >>>>>>> asterisk-users mailing list<br>
> >>>>>>> To UNSUBSCRIBE or update options visit:<br>
> >>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users<br>
> >>>>>>><br>
> >>>>><br>
> >>>><br>
> >>><br>
> >><br>
> ></span></p>
</div>
<br>--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users </body>
</html>