hi: maybe you can try noload res_timing_timerfd in modules.conf and see what asterisk pick up for timing. in my system, if I disable res_timing_timerfd, then dahdi timing is selected and system become stable.
Regards, tbskyd 2011/5/14 satish patel <satish...@hotmail.com>: > You mean say i don't use res_timing_dahdi.so ? I guess this is just timing > module nothing related to Card. > > _S > > ________________________________ > From: tu...@canistec.com > Date: Fri, 13 May 2011 18:30:52 +0200 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] 1.8 and prematuremedia problem > > sangoma cards do not use dahdi... > > 13.5.2011 v 17:16, satish patel <satish...@hotmail.com>: > > 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 ? > > campbx1*CLI> module show like timing > Module Description Use > Count > res_timing_pthread.so pthread Timing Interface > 0 > res_timing_timerfd.so Timerfd Timing Interface > 1 > res_timing_dahdi.so DAHDI Timing Interface > 0 > 3 modules loaded > > > ________________________________ > From: n...@njcolledge.net > To: asterisk-users@lists.digium.com > Date: Fri, 13 May 2011 15:11:19 +0000 > Subject: Re: [asterisk-users] 1.8 and prematuremedia problem > > At the asterisk CLI type “module show like timing” > > > > Whichever has a use-count >1 is the one you are using. > > > > Nic. > > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel > Sent: 13 May 2011 16:03 > To: tbs...@gmail.com; asterisk-users > Subject: Re: [asterisk-users] 1.8 and prematuremedia problem > > > > Thanks for reply, > > How do i find asterisk using which timing res_timing_timerfd or > res_timing_dahdi ? > > -S > >> Date: Fri, 13 May 2011 22:13:47 +0800 >> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem >> From: tbs...@gmail.com >> To: satish...@hotmail.com; asterisk-users@lists.digium.com >> >> 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? 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 >> >>>>>>>>>>>> >> >>>>>>>>>>> >> >>>>>>>>>>> -- >> >>>>>>>>>>> >> >>>>>>>>>>> >> >>>>>>>>>>> >> >>>>>>>>>>> _____________________________________________________________________ >> > >> > >> >>>>>>>> >> >>>>>>>> >> >>>>>>>>>>> -- Bandwidth and Colocation Provided by >> >>>>>>>>>>> http://www.api-digital.com >> >>>>>>>>>>> -- >> >>>>>>>>>>> New to Asterisk? 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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 >> >>>>>>> >> >>>>> >> >>>> >> >>> >> >> >> > > > -- _____________________________________________________________________ -- > 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 > > -- > _____________________________________________________________________ > -- 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 > > -- _____________________________________________________________________ -- > 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 > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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