hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk.
Regards, tbskyd 2011/5/13 Satish Patel <satish...@hotmail.com>: > Glad you solved it. Now I'm having high CPU load issue. I don't know why but > sometime my asterisk process reached ~150% CPU load and just locked no calls > nothing only solution is kill -9 > > I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because > of low through put ?? Which OS are you using? > > -- > Sent from my iPhone > > On May 12, 2011, at 9:31 PM, d tbsky <tbs...@gmail.com> wrote: > >> hi: >> sorry. the issue number is 19268. not 19628. >> sorry about that!! >> >> Regards, >> tbskyd >> >> 2011/5/13 d tbsky <tbs...@gmail.com>: >>> >>> hi: >>> I report my issue as issue 19628. >>> it is fixed and I run asterisk 1.8 in production now. >>> thanks a lot for your help! >>> >>> Regards, >>> tbskyd >>> >>> 2011/5/11 d tbsky <tbs...@gmail.com>: >>>> >>>> hi: >>>> ok I will create a bug report. and I found I still need >>>> "prematuremedia=no" in asterisk 1.6.2.18. >>>> yesterday I was testing at home with zoiper softphone + iax. today I >>>> test snom hardware sip phone and found that "prematuremedia=no" is >>>> still necessary. >>>> >>>> Regards, >>>> tbskyd >>>> >>>> >>>> 2011/5/11 satish patel <satish...@hotmail.com>: >>>>> >>>>> I am sorry about that but its interesting it doesn't work with 1.8 SVN >>>>> >>>>> I would say please report this bug so that way you can track issue, And >>>>> may >>>>> be in future it help us :) >>>>> >>>>> -S >>>>> >>>>>> Date: Wed, 11 May 2011 01:31:34 +0800 >>>>>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem >>>>>> From: tbs...@gmail.com >>>>>> To: asterisk-users@lists.digium.com; satish...@hotmail.com >>>>>> >>>>>> hi: >>>>>> that issue is marked as fixed, so no more comment can be added :( >>>>>> anyway, I try the following combination: >>>>>> 1.8.3.2 + sig_pri patch >>>>>> 1.8 svn which already has sig_pri patched >>>>>> 1.8.4 + libpri patch (another unofficial patch in issue 18868) >>>>>> >>>>>> but none works. >>>>>> >>>>>> finally I downgrade to 1.6.2.18 and I found everything works. I don't >>>>>> even need to set "prematuremedia" with 1.6.2.18. >>>>>> so I think I will need to stay with 1.6.2 a little longer... >>>>>> >>>>>> thanks a lot for your help!! >>>>>> >>>>>> Regards, >>>>>> tbskyd >>>>>> >>>>>> 2011/5/10 satish patel <satish...@hotmail.com>: >>>>>>> >>>>>>> Also i would say add comment on following issue if after patch you >>>>>>> having >>>>>>> issue, That way it help community to fine tune patch. >>>>>>> >>>>>>> https://issues.asterisk.org/view.php?id=18868 >>>>>>> >>>>>>> Good luck >>>>>>> >>>>>>> >>>>>>>> From: satish...@hotmail.com >>>>>>>> To: tbs...@gmail.com >>>>>>>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem >>>>>>>> Date: Tue, 10 May 2011 07:43:47 -0400 >>>>>>>> CC: asterisk-users@lists.digium.com >>>>>>>> >>>>>>>> I have applied this patch in 1.8 svn branch and it works great for >>>>>>>> me. >>>>>>>> >>>>>>>> I have nothing special configuration just simple dial command for >>>>>>>> outgoing call. >>>>>>>> >>>>>>>> Also check there are progress=yes option in chan_dahdi >>>>>>>> >>>>>>>> -- >>>>>>>> Sent from my iPhone >>>>>>>> >>>>>>>> On May 10, 2011, at 5:58 AM, d tbsky <tbs...@gmail.com> wrote: >>>>>>>> >>>>>>>>> hi: >>>>>>>>> I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not >>>>>>>>> apply to 1.8.3.2 or 1.8.4-rc3). >>>>>>>>> but the situation is the same. do I need to play with other options >>>>>>>>> with the patch? or I need >>>>>>>>> newer asterisk versions to solve the problem? >>>>>>>>> thanks a lot for information!! >>>>>>>>> >>>>>>>>> 2011/5/10 d tbsky <tbs...@gmail.com>: >>>>>>>>>> >>>>>>>>>> hi: >>>>>>>>>> thanks a lot for your quick reply. I saw that patch and think that >>>>>>>>>> it was already included in 1.8.3. >>>>>>>>>> now I know it will be included in 1.8.5. >>>>>>>>>> I will try it and thanks again for your kindly help!! >>>>>>>>>> >>>>>>>>>> 2011/5/10 Satish Patel <satish...@hotmail.com>: >>>>>>>>>>> >>>>>>>>>>> Apply this patch https://issues.asterisk.org/view.php?id=18868 >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> Sent from my iPhone >>>>>>>>>>> >>>>>>>>>>> On May 9, 2011, at 9:57 PM, d tbsky <tbs...@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> hi: >>>>>>>>>>>> our current connection is below: >>>>>>>>>>>> >>>>>>>>>>>> sip phone<--->asterisk<---->alcatel PBX<---->PSTN >>>>>>>>>>>> >>>>>>>>>>>> asterisk and alcatel PBX is connected via E1 isdn-pri. >>>>>>>>>>>> >>>>>>>>>>>> when I use sip phone to dial outside PSTN world: >>>>>>>>>>>> 1. with 1.4 it is fine. >>>>>>>>>>>> 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or >>>>>>>>>>>> sip >>>>>>>>>>>> phone can not hear the ring and the beginning of the PSTN voice. >>>>>>>>>>>> 3. with 1.8.3.2, I can not hear ring and the beginning of the >>>>>>>>>>>> PSTN >>>>>>>>>>>> voice. I try to play options with "prematuremedia" and >>>>>>>>>>>> "progressinband". but I can not find working settings. >>>>>>>>>>>> >>>>>>>>>>>> I don't know what other options I can try. >>>>>>>>>>>> thank a lot for information!! >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _____________________________________________________________________ > > >>>>>>>> >>>>>>>> >>>>>>>>>>>> -- Bandwidth and Colocation Provided by http://www.api- >>>>>>>>>>>> digital.com -- >>>>>>>>>>>> New to Asterisk? 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