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?

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On May 12, 2011, at 9:31 PM, d tbsky <tbs...@gmail.com> wrote:

hi:
  sorry. the issue number is 19268. not 19628.
  sorry about that!!

Regards,
tbskyd

2011/5/13 d tbsky <tbs...@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 <tbs...@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...@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: tbs...@gmail.com
To: asterisk-users@lists.digium.com; satish...@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...@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...@hotmail.com
To: tbs...@gmail.com
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
Date: Tue, 10 May 2011 07:43:47 -0400
CC: asterisk-users@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

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On May 10, 2011, at 5:58 AM, d tbsky <tbs...@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 <tbs...@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...@hotmail.com>:
Apply this patch https://issues.asterisk.org/view.php? id=18868

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On May 9, 2011, at 9:57 PM, d tbsky <tbs...@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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