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Error when trying to add Office 365 Calendar

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Hi,

I update our FreePBX 13 to FreePBX 14 and I'm getting this error when I try to add an Office 365 Calendar

The message appears when FreePBX is attempting to load the calendar list (after I click autodetect)
I tried to look in the different logs but I didn't find anything.

Any ideas or leads ?


Lots of Bad Dest after Upgrade to 14

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That seemed to work.. Thanks you..

Error when trying to add Office 365 Calendar

Cisco ATA186 problem

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... and other Cisco devices have even more limited passwords. Some of the older phones, for example, are limited to about 10 characters.

DTMF + BLF with Yealink phones?

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Thanks. We ended up showing them how to transfer that way

Upgrade 13 to 14?

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do you guys recommend upgrading to 14? I don't see it as an option with I run the CLI command to switch versions.

Upgrade 13 to 14?

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Asterisk or FreePBX?

On my FreePBX 14 system I can switch to Asterisk 14... I don't believe it's possible with FreePBX 13...

Have a nice day!

Nick

Video call does not have video

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Video calling was working on all our version 13 systems, now video and call history/CDR isn't working after upgrading to 14.

Video calling is also not working on clean installs of version 14, however, call history/CDR is working on in version 14 on clean version 14 installs (not upgraded from an older version).


UCP data stopped after upgrade to FreepBX 14

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Thanks the rename worked. Now if I can just get video working in 14.

Adding Custom iptables rule to FreePBX Firewall

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I need to add a custom iptable rule to allow postrouting NAT for IPSEC tunnels to our PBX, but haven't figured out how to do so in a way that doesn't get overwritten.

I know it's bad form, but I even added the command to rc.local, but the firewall sometimes takes a while to load and then overwrites that.

Could you please share the specifics of how you added custom firewall rules to the FreePBX firewall module that are persistent even after a reboot?

Thanks!

see: https://wiki.freepbx.org/display/FPG/Firewall

UCP data stopped after upgrade to FreepBX 14

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Hi Russell!

Interesting... Which version of the "script" did you run?

I wish I could help you there but I never even attempted to have video...

Try posting some logs of an attempt to get this working, maybe there's something in there that would ring a bell to someone...

Good luck and have a nice day!

Nick

Video call does not have video

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Hi!

Russell got it working by doing this:

As for your video problem @russman you might want to post some logs... Maybe someone will be able able to help you...

I doubt I will since I never even attempted that but hopefully someone will...

Good luck and have a nice day!

Nick

Error when trying to add Office 365 Calendar

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Thank you for your answer.

Actually, i didn't add it as an iCal calendar, I chose Remote Outlook. I typed in my Office 365 user and password and the EWS Server URL was correctly detected.

If I have to chose an ICS calendar (which I will try tomorrow), what does "Remote Outlook" mean ?

Upgrade 13 to 14?

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I just upgraded from 13 to 14 las night. A few hiccups but so far everything is good.

List of some problems I encountered, nothing too serious:
- fail2ban went on a rampage and blacklisted everything on the LAN that was trying to acces web interface
- I had to reconfigure some email settings
- Some options we're showing enabled while in fact they we're disabled (I didn't take note of which ones, it was getting late)
- I had to reinstall a few modules

Zulu Connection dropped by remote peer

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Hi TM1000

SMS 13.0.11.3 Stable Sangoma Technologies Corporation Commercial Enabled

It was installed long ago.


Maxptime = 150

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I would like to know more about how to change the ptime for PJSIP too. Seems setting ulaw:10 or g722:10 does nothing.

Maxptime = 150

CallerID By Physical Location, not Area Code?

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Hi!

No...

About 20 years ago a place where I worked used to do something pretty similar...

They would take the same area code + Central Office/Exchange as the main phone number + the 4 digit extension number.

Now, for extensions which were externally reachable (back then numbers beginning with 8 and later 7) this was a legit phone number which they owned but for all the other extensions it was a number they did not own...

When I said this was not OK and asked why I was told that this was done because some people (which owned them money I believe if this was done for the department I think) would filter out their calls if they saw the generic phone number of the company...

Now it wasn't very legit to do what they did but the actual intent was not to commit a crime but to reach customers which were failing to meet their obligations towards them...

There was nothing I could do about it (beside saying this was not ok) since I was just normal IT staff...

They eventually stopped doing that...

Have a nice day!

Nick

Intrusion detection blocks by home ip

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It just blocked my home ip again. I sshed into it and it shays:

Broadcast message from root@pbx.domain.com
Firewall service now starting.

What could cause this issue?

PJSIP NAT AUDIO transport=transport-udp

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I have a server and a remote endpoint both behind NAT. I am having audio issues and see that the sdp message is showing the phone ip not the server. In reading the asterisk documentation it suggests having the "transport=transport-udp" enabled in the endpoint context. I didn't see that option for the endpoint. Would anyone have any suggestions?

;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
348 ;
349 ; This example assumes your transport is configured with a public IP and the
350 ; endpoint itself is behind NAT and maybe a firewall, rather than having
351 ; Asterisk behind NAT. For the sake of simplicity, we'll assume a typical
352 ; VOIP phone. The most important settings to configure are:
353 ;
354 ; * direct_media, to ensure Asterisk stays in the media path
355 ; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
356 ;
357 ; Depending on the settings of your remote SIP device or NAT/firewall device
358 ; you may have to experiment with a combination of these settings.
359 ;
360 ; If both Asterisk and the remote phones are a behind NAT/firewall then you'll
361 ; have to make sure to use a transport with appropriate settings (as in the
362 ; transport-udp-nat example).
363 ;
364 ;[6002]
365 ;type=endpoint
366 ;transport=transport-udp
367 ;context=from-internal
368 ;disallow=all
369 ;allow=ulaw
370 ;auth=6002
371 ;aors=6002
372 ;direct_media=no
373 ;rtp_symmetric=yes
374 ;force_rport=yes
375 ;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
376 ;ice_support=yes ;This is specific to clients that support NAT traversal
377 ;for media via ICE,STUN,TURN. See the wiki at:
378 ;https://wiki.asterisk.org/wiki/x/D4FHAQ
379 ;for a deeper explanation of this topic.
380
381 ;[6002]
382 ;type=auth
383 ;auth_type=userpass
384 ;password=6002
385 ;username=6002
386
387 ;[6002]
388 ;type=aor
389 ;max_contacts=2
390
391

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