[asterisk-users] 1.8 and prematuremedia problem
Satish Patel
satish_lx at hotmail.com
Sun May 15 09:38:13 CDT 2011
Thanks and I did that and my figure are cross now. Let see
--
Sent from my iPhone
On May 15, 2011, at 8:35 AM, d tbsky <tbskyd at gmail.com> wrote:
> hi:
> maybe you can try noload res_timing_timerfd in modules.conf and see
> what asterisk pick up for timing.
> in my system, if I disable res_timing_timerfd, then dahdi timing is
> selected and system become stable.
>
> Regards,
> tbskyd
>
> 2011/5/14 satish patel <satish_lx at hotmail.com>:
>> 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!!
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> --
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _____________________________________________________________________
>>>>
>>>>
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>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
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>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>
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>>>>>>>>>> _____________________________________________________________________
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>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>
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>
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