Hello Miles,

from mobile (sorry for typos ;)


On Fri, May 13, 2022, 14:23 Ali Alhaidary <[email protected]>
wrote:

> Hi,
>
> Very good point #2, can we change the default installation folder to be
> "OpenMeetings" and the old one (if there is one) "OpenMeetings.old" or
> give the user the option to chose the extension he wants, in the
> installation manual.
>
> Also, lets separate data folder from the app folder, we usually do that
> manually so that pointers are not broken...
>
>
> Ali
>
> On 5/12/22 20:02, Yah's Global Kingdom wrote:
> > Hi,
> >
> > 1.  1st  let me applaud you on the drastic improvement in performance
> > and stability between 504 and 620.  Hats off to you all
>

Thanks! :)

>
> > I just upgraded from Open504 to Open620 using this
> > https://openmeetings.apache.org/Upgrade.html as a guide.
> >
> > I created a OM-backup and  created a mysql backup.  I have large files
> > wanted to see if this was fixed as well.
> >
> > 2.  There were some issues restoring from the backup. The importer was
> > unsuccessful in importing videos and images. It was not able to
> > successful convert them as path to the video was pointing to the old
> > instance of OM, which had been rename to open504.bak.  But the
> > importer was looking for the files in the old location.  I basically
> > had to truncate om_file, file_log and invitations tables to remove the
> > old links.  The restore from the mysql backup put all the other
> > configuration and user information back in place.
> >
> > A fix for this may  be to include in the upgrade instructions to
> > change the name of the old OM installation back to the original name
> > before importing the OM backup into the new installation.
>

Hmm
Usually backup includes all files/recordings

Maybe you *exclude* files while backup?

If no
Can you please share some example of such backup? (You can do it privately
:)


> >
> > 3.  I completely reinstalled Asterisk 16.
> > Purchase a DID and I am able to dial out from the asterisk box to the
> > PTSN and to SIP address.  However, I am unable to get the SIP dialer
> > to do anything and I am unable to dial into any conference room.  I do
> > a podcast and the goal is to be able to dial into the podcast using
> > the SIP dialer. I can dial out from extensions, I have created but I
> > can not any with the sip dialer.
> >
> > It would seem that OMSIP records would need AOR, AUTH, AND ENDPOINT
> > records in Asterisk for the dialer to work. Does anyone have a working
> > SIP dialer configuration for Asterisk or that can look at the document
> > that I have attached of my configurations.  I will  better document
> > this process and return it to the community for anyone else that wants
> > to do the same or similar thing.  Right now I am just trying to get
> > the SIP Dialer to work and to be able to make calls using
> > OpenMeetings.  Thanks ahead of time.  OH in the attached file is log
> > output when the SIP Dialer is Initiated, the Call button is pressed
> > and when the SIP Dialer is closed.  That is all the output I could
> > find in the logs.  Also as I followed
> > https://openmeetings.apache.org/AsteriskIntegration.html I didn't
> > include all the configurations in that document but most of
> > them,  including those needed to configure a working incoming outgoing
> > extension to the PSTN from the  ITSP and to create working internal
> > extensions in Asterisk that are able to dial out to the PSTN.
> >
> >
> > Again my goal is to be able to dial out from OM to my podcast or have
> > people be able to dial into OM conference and also listen and
> > participate in the podcast.
>

I can try to help here
BUT unfortunately I can test only incoming soft-phone calls locally

Please share how can i do outgoing call
And I'll try to reproduce your steps :)


>
> > Thanks ahead of time.
>

BTW 6.3.0 is currently being voted :)

>
> > Miles
> >
> >
> >
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