[asterisk-users] 1.8 and prematuremedia problem

Fri May 13 09:13:47 CDT 2011

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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