[asterisk-users] 1.8 and prematuremedia problem
turby canistec
turby at canistec.com
Fri May 13 11:30:52 CDT 2011
sangoma cards do not use dahdi...
13.5.2011 v 17:16, satish patel <satish_lx at 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: nic at njcolledge.net
> To: asterisk-users at 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-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel
> Sent: 13 May 2011 16:03
> To: tbskyd at 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: tbskyd at gmail.com
> > To: satish_lx at hotmail.com; asterisk-users at 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_lx at 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 <tbskyd at gmail.com> wrote:
> > >
> > >> hi:
> > >> sorry. the issue number is 19268. not 19628.
> > >> sorry about that!!
> > >>
> > >> Regards,
> > >> tbskyd
> > >>
> > >> 2011/5/13 d tbsky <tbskyd at 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 <tbskyd at 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_lx at 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: tbskyd at gmail.com
> > >>>>>> To: asterisk-users at lists.digium.com; satish_lx at 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_lx at 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_lx at hotmail.com
> > >>>>>>>> To: tbskyd at gmail.com
> > >>>>>>>> Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
> > >>>>>>>> Date: Tue, 10 May 2011 07:43:47 -0400
> > >>>>>>>> CC: asterisk-users at 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 <tbskyd at 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 <tbskyd at 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_lx at 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 <tbskyd at 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!!
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> --
> > >>>>>>>>>>>>
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> _____________________________________________________________________
> > >
> > >
> > >>>>>>>>
> > >>>>>>>>
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> > >>
> > >
>
>
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