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? 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