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 <[email protected]> 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 <[email protected]>:
You mean say i don't use res_timing_dahdi.so ? I guess this is
just timing
module nothing related to Card.
_S
________________________________
From: [email protected]
Date: Fri, 13 May 2011 18:30:52 +0200
To: [email protected]
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
sangoma cards do not use dahdi...
13.5.2011 v 17:16, satish patel <[email protected]>:
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: [email protected]
To: [email protected]
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: [email protected]
[mailto:[email protected]] On Behalf Of
satish patel
Sent: 13 May 2011 16:03
To: [email protected]; 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: [email protected]
To: [email protected]; [email protected]
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 <[email protected]>:
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 <[email protected]> wrote:
hi:
sorry. the issue number is 19268. not 19628.
sorry about that!!
Regards,
tbskyd
2011/5/13 d tbsky <[email protected]>:
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 <[email protected]>:
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 <[email protected]>:
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: [email protected]
To: [email protected]; [email protected]
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 <[email protected]>:
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: [email protected]
To: [email protected]
Subject: Re: [asterisk-users] 1.8 and prematuremedia problem
Date: Tue, 10 May 2011 07:43:47 -0400
CC: [email protected]
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 <[email protected]>
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 <[email protected]>:
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 <[email protected]>:
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 <[email protected]>
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!!
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-
digital.com --
New to Asterisk? Join us for a live introductory webinar
every
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by
http://www.api-digital.com
--
New to Asterisk? Join us for a live introductory
webinar every
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com
--
New to Asterisk? Join us for a live introductory webinar
every
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
--
Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To
UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
--
Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To
UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users