[asterisk-users] upgrade asterisk 11 to 13 or 11-16

[Jöran Vinzens]

Hi,

Using SIPp to check your asterisk is working has some pitfalls.
We recorded SIP invites (as these are the important parts of a call) of a
normal call going through the Asterisk. We recorded it the old working way.
In General we just compared what we had before with what we have after the
upgrade. So if the Headers are Correct (from, to, our own X header, P
Header etc). Since there are a couple of things you might want to check,
e.g. someone is not allowed to place long distance calls, you can check the
behavior of the asterisk as well so if the calls get rejected when some
user dials some number.

In SIPp there are these Scenario Files (XML Files) that contain a sequence
of SIP Messages to send/receive. Using the receiving of Messages you can
specifically check for presence or absence of a Header or a field in a
header.

there are lots of examples in the github repo
https://github.com/SIPp/sipp

For a A calls B call you need to start two SIPp Instances (one sending the
call, one receiving the call)
If your clients register to your Asterisk no not forget to do so, otherwise
the Asterisk has no AOR to forward the call to. (Using plain UDP helps here
a lot).

The first check you build up might be some more work even if you never
played around with SIPp but all what follows are quite simple and ensure
quality.

for my talk i put everything together you might need to place a simple call
to an asterisk
https://github.com/sipgate/signaling-test

the start shell script will start first a registration to your Asterisk and
then starts two sipp instances to place the call.

feel free to use it.

BR
Jöran

On Mon, Dec 7, 2020 at 8:29 PM Eric Wieling <ewieling at nyigc.com> wrote:

> I'm sure you can, but I've never done it.
>
> On 12/7/20 2:18 PM, thelma at sys-concept.com wrote:
> > Sound reasonable.  I know it take time to debug the dial-plan after
> upgrade.
> >
> > Can I use sipp, from command line to call my local asterisk specific
> > extension and to observe in another terminal via "asterisk -vvvvvvr"
> > what it is doing?
> >
> >
> > On 12/07/2020 11:50 AM, Eric Wieling wrote:
> >> Read UPGRADE.TXT in v13 and v16.  Then read it again.
> >>
> >> I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were
> >> resolved, then I switched to PJSIP.   Once all the issues with PJSIP
> >> were resolved, then I upgraded from v13 to Asterisk v16.   This was done
> >> over the course of about a year, but I was not in any hurry.
> >>
> >> PJSIP configuration is fundamentally different chan_sip configuration. I
> >> don't recommend switching to PJSIP and upgrade Asterisk at the same
> time.
> >>
> >> On 12/6/20 3:38 PM, thelma at sys-concept.com wrote:
> >>> I'm planning to upgrade my asterisk-11.25 to ver. 13
> >>> or should I go to 11 to 16
> >>>
> >>> Is there any official documentation how to upgrade, what to watch for
> >>> during upgrade?
> >>>
> >>>
> >>
> >
>
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>
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-- 

Jöran Vinzens - vinzens at sipgate.de
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Telefax: +49 211-63 55 55-22

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