[asterisk-users] Asterisk 1.8.3

[Satish Patel]

Right now I'm testing 1.8.3 in devlopment and respose it pretty good  
without realtime. (I didn't set realtime).

I ran stress test with sipp and pass 5000 call with RTP and no issue  
at all. I got hogging at system resource but no issue at asterisk.

Look like I might go with 1.8.3 and later upgrade with 1.8.4 asap.

--
Sent from my iPhone

On Apr 7, 2011, at 9:12 AM, "Bryant Zimmerman" <BryantZ at zktech.com>  
wrote:

>
>
>
> On Apr 7, 2011, at 8:51 AM, Ishfaq Malik <ish at pack-net.co.uk> wrote:
>
> > On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
> >>
> >> On Apr 6, 2011, at 8:54 PM, Edwin Lam <edwin.lam at officegeneral.com>
> >> wrote:
> >>
> >>> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
> >>>>
> >>>> Thanks for your response. I have added the patch for 18818 per
> >>>> Michel Verbrask's
> >>>> recomendation. It appers that it has made quite a difference. I
> >>>> don't have an PRI
> >>>> connections as all of our PRI's are connected via SIP gateways. I
> >>>> did run into
> >>>> serveral instances wher I had to kill -9 the process as well but
> >>>> post patch I have
> >>>> been in good shape know on wood. I hope there will be a new
> >> release
> >>>> that will
> >>>> address the stability issues very soon if they release 1.8.4
> >>>> without cleaning this
> >>>> up I won't move unitl it is addressed.
> >>>
> >>> looking back at the messages file for the past 2 days. it
> >>> just hanged on totally different events none of which related
> >>> to Local channels.
> >>>
> >>> as far as the PRI not hearing early media issue. here's the
> >>> excerpt from the messages file after "pri debug on" command:
> >>>
> >>> *********************
> >>>
> >>> -- Executing [18008291011 at out_going_x:1] Dial("SIP/
> >>
> >> ... Parts Removed see origional response
> >>
> >>> -- Processing IE 30 (cs0, Progress Indicator)
> >>> -- PROGRESS with cause code 127 received
> >>> -- DAHDI/34-1 is making progress passing it to SIP/4988-6-00000b45
> >>>
> >>> ***********************************
> >>>
> >>> i used the same SIP station to dial the same 800 number
> >>> on both versions (1.8.3.2 & 1.6.2.17). the output are
> >>> pretty much identical except on 1.8.3.2, after the
> >>> "PROGRESS with cause code 127..." message. i would hear
> >>> nothing until the other side timed out & hang up, whereas on
> >>> 1.6.2.17. i got the "DAHDI/... is making progress passing it to
> >>> SIP..."
> >>> message and can hear the early media from the other side.
> >>>
> >>>
> >>>> For Now 1.8.3..2 is very bad.
> >>>
> >>> agreed...
> >>
> >> From: "Satish Patel" <satish_lx at hotmail.com>
> >> Sent: Thursday, April 07, 2011 8:22 AM
> >> Oh! Boy,
> >>
> >> Is it ture 1.8.3 is unstable? We are planning to put this in
> >> production. Please suggest me what should I do?
> >>
> >>
> >> Satish
> >>
> >> For me 1.8.3.2 has been the worst build that I have tried to use as
> >> far a stability in a very long time. We are having issues
> >> with deadlocks and voicemail.
> >> I don't have a good option for you if you want to run 1.8 currently
> >> the most stable release version I have found is 1.8.2.3 but I am
> >> having the Voicemail issues there as well.
> >> Things like messages not deleting propperly and hanging up the mail
> >> box so users can't check them.
> >
> > 1.8.2 is unusable if you use RealTime without the patch in this  
> issue
> >
> > https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
> >
> >
>
>  From: "Satish Patel" <satish_lx at hotmail.com>
> Sent: Thursday, April 07, 2011 9:06 AM
>
> We don't have realtime configuration everything is in plain text file.
>
> Is 1.8.3 has realtime issue or general issue?
>
> Satish
> I have seen my issues with the realtime disabled and using just  
> plain text. The issues get worse for me when we move to our realtime  
> confgs. So from my perspective I would say you might get farther  
> with realtime off but I would not bank on it.
>
>
> --
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