[asterisk-users] 1.8 and prematuremedia problem

satish patel satish_lx at hotmail.com
Fri May 13 10:16:02 CDT 2011


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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> >

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> >>>

> >>

> >




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