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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>> ><br> </body>
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