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Did the upgrade to Asterisk 11.21.2 and found out that the new sounds were not there either. Did the manual download from downloads.asterisk.org/pub/telephony/sounds/ Definitely a big improvement.

Same here. I just upgraded to Asterisk 11.21.2 and all records before the upgrade show up in CDR, nothing after the upgrade however. When looking at the tail of Master.csv it did record all calls made/received after the upgrade.

Did you check in extensions_custom.conf whether the extension is assigned there?

As long as both devices are connected to the same user they should both ring. This is done by selecting the same default user at the bottom of the device configuration.

My desk phones automatically connect to phone server and prompted for device username and password. On iOs I use Groundwire and edit username and password in the account settings. Assume editing the phone through webinterface will work.

If you go to the applications menu in FreePBX and select devices then you can add devices without the need for MAC. It generates a password that you have to enter on the device for it to register.

tried to manually install asterisk and run into an issue on Asterisk-Flite when I run the "make" command the output is gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_flite.o app_flite.c app_flite.c: In...

Tried to reinstall Asterisk using Ward's script and got "Asterisk died with code 127"

Have you tried using one of their regional servers rather than the main? I have two trunks with them one registered on main and other on regional proxy.

I actually have rtpkeepalive at 20. I a very distant past (way before I noticed the current issue) had it at 0 but increased to 20 when I had a registration issue.

For some time now calls on my system get dropped when the line is on mute (e.g. to listen in on conf calls). The system gives me a hangup cause 16 which is normal clearing. When evaluating the call length it turns out they are all around the 300 seconds which happens to be the rtpholdtimeout...

The easiest way to access the web page of Piaf is to set up a VPN connection. That way you will be regarded as an internal user. Otherwise you would have to open port 80 on the Draytek as well. Are you running incrediblepbx? In that case you have to allow connections from the external IP...

On the Draytek instead of using filters configure "Open Ports" in the NAT settings.

Did you set up the port forwarding from your router to your PBX correctly?

lgaetz, fair point and had not though about that. My backup provider just replied that they need my address so that emergency services will know from where I am calling.

My VOIP provider does not provide support for emergency numbers. I live in the UK and want my users to be able to connect to 999 in case of an emergency. Does anyone know an alternative way to connect to the emergency number?

Does anyone have the Aastra scripts running under FreePBX 12?

Actually ignore previous message, message is "BASH vulnerability not found"

Had the same problem and retrying does not solve the issue. The error remains. Also tried a server reboot, but still the same.

Can anyone in the forum confirm whether the FreePBX upgrader (think current version is RC9) has worked for them without breaking PIAF?