[asterisk-users] 1.8 and prematuremedia problem
satish patel
satish_lx at hotmail.com
Fri May 13 11:42:21 CDT 2011
You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card.
_S
From: turby at canistec.com
Date: Fri, 13 May 2011 18:30:52 +0200
To: asterisk-users at 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_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!!
> >>>>>>>>>>>>
> >>>>>>>>>>>> --
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>> _____________________________________________________________________
> >
> >
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>>>>> -- 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
--
_____________________________________________________________________
-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110513/74370034/attachment-0001.htm>
More information about the asterisk-users
mailing list