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.
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.
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.
Thanks ahead of time.
Miles
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