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Posted to user@openmeetings.apache.org by Yah's Global Kingdom <ya...@gmail.com> on 2022/05/12 17:02:12 UTC
Upgrade fromOpen504 to Open620
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
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
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Re: Upgrade fromOpen504 to Open620
Posted by Maxim Solodovnik <so...@gmail.com>.
And
This email should go user@ list :)
from mobile (sorry for typos ;)
On Fri, May 13, 2022, 23:41 Maxim Solodovnik <so...@gmail.com> wrote:
> Hello Miles,
>
> from mobile (sorry for typos ;)
>
>
> On Fri, May 13, 2022, 14:23 Ali Alhaidary <al...@the5stars.org>
> 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
>> >
>> >
>> >
>> > <
>> https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
>>
>> > Virus-free. www.avast.com
>> > <
>> https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>>
>> >
>> >
>
>
Re: Upgrade fromOpen504 to Open620
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello Miles,
from mobile (sorry for typos ;)
On Fri, May 13, 2022, 14:23 Ali Alhaidary <al...@the5stars.org>
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
> >
> >
> >
> > <
> https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
>
> > Virus-free. www.avast.com
> > <
> https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>
> >
> >
Re: Upgrade fromOpen504 to Open620
Posted by Ali Alhaidary <al...@the5stars.org>.
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
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
> Virus-free. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
>
>