Who is also running an Asterisk phone server as a hobby or is planning to do so?

[markosjal]

If your FreePBX Pi is behind a firewall (router) and you do not open ports to the public Internet, just make sure you change any default passwords and you will be perfectly fine. The major security problems arise when people open ports to the Internet.

You only really need to open ports on your router/firewall if you have any of the following

• Improperly configured sip.conf
• Phones located off site
• Access to the Web GUI from a remote site
• more

If yoou do need remote web GUI access, Sometimes it is way better to use a remote desktop on a system (even if on the same Pi) and get web GUI access remotely, that way. It is not as responsive but more secure. When I have ran GUI configurations that were open to the Internet in the past I always disabled the web server unless I really needed it

NO SIP or RTP ports need be open at all if not using off site phones.

[dsk]

I have now tried to configure freepbx and ended up with this unsolvable....

[dsk]

I just started from scratch.. Now it seems to be working pretty well

[markosjal]

So glad you got through it.

Which install did you end up  doing?

[dsk]

FreePBX 15.0.29  after running som updates. I downloaded the raspbx-10-10-2020 from this page: http://www.raspberry-asterisk.org/downloads/

I had to use a keboard and screen to chang the root password, and set the right time zone. Setting time zone in Freepbx alone was not enough to get everything right.

You may call  17772922141149@in.callcentric.com or just 17772922141149 from Callcentric, and get the Norwegian time. (in English) 

https://www.callcentric.com/faq/4/130

[Volker]

You may call  17772922141149@in.callcentric.com or just 17772922141149 from Callcentric, and get the Norwegian time. (in English) 

It works!

[markosjal]

Yep! 22:33 and 0 Seconds was me!

I have had direct dialing of call centric numbers for many years on my asterisk and rarely used it, until recently when Callcentric deleted my nearly 20 year old account because they did not like me using them to do a "transitional port" of a problematic number port.

Since in.callcentric.com does not work too well from behind a NAT , I set up a SIPBROKER trunk for NAT then I can use that trunk for anything on sipbroker with the proper prefix (*462 to sipbroker) .

Works as expected, I can dial direct to the +1777 Callcentric numbers again.  !

[markosjal]

DSK,
It looks like I already allocated you a Boston DID Number , so let me know if you need it redirected. It is currently directed to a localphone account.

[dsk]

The Boston number works great as it is Thank you!

[markosjal]

Yea the bummer about dialing those call centric LONG numbers is that if you delay on the last three digits you connect to the main account. Sorry about that

[dsk]

Changed the time schedule, so after 22:30 everything goes to the clock No-one leaves messages, so that may be a hint

[markosjal]

I never thought about that send late night callers to a talking clock, what a great idea!

[dsk]

Hi,markosjal I need to ask for help, agin.
I tried to setup the ata for new numbers, but I am not able to do it right.
The password is right, The numbers are 41 and 42, and the IP of freePBX is 192.168.1.111
Are you able to see what may be wrong?

Thank you
dsk

[dsk]

It is something with the FreePBX, and not the ATA.
Thank you!

[markosjal]

What  do asterisk logs say?

You should be able to load up the asterisk console with

asterisk -vvvvvvvvvvvvr

then apply power to the ATA and at some point you will see a couple of lines about

....register extension ...... (it may say password or the extension does not exist )

if it never mentions the registration attempt you  may be registering to the wrong IP or Port. with newer versions of Asterisk this is particularly important as there are two SIP stacks (Chan_SIP and PJSIP) and each runs on a different port. The last version of FreePBX I say years ago you had to select the port and primary SIP stack (PJSIP or CHAN_SIP). Confirm port number(s) used for SIP and PJSIP

The second image you sent is irrelevant for SIP . It only applies to MGCP

Also you may want to check NAT settings as I have seen incorrect NAT settings cause registration failures.

Also please post your Chan_SIP and PJSIP bind settings (port numbers). I strongly suspect you are registering to the wrong port

In the UTSTARCOM ATA I can see a "Proxy Port" of 5060 but then you have an "outbound proxy Port" set to 5065


Also... disable VAD (in last image) as it most never works in asterisk, and unless you have specifically installed G729 codec you probably want default codec at g711A ; A.K.A. ALAW (in Europe) or g711u A.K.A. ALAW (in N. America) . In this day and age of crappy cell phone calls being the norm it is best to stick to G711A/U and avoid microcodecs altogether like G729, G723, G726, GSM otherwise too much transcoding can occur in a call leading to calls being heard but not intelligible (Wa Wa Wa)  like this https://www.youtube.com/watch?v=ss2hULhXf04&t=14s