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
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