[asterisk-users] 1.8 and prematuremedia problem

Satish Patel satish_lx at hotmail.com
Fri May 13 07:11:18 CDT 2011 

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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