TRY THIS IPBX 13-13.10 Incoming Routes Setup

[Milton Alvis]

Newbe to IPBX 13-13.10:
SiP.US trucks and Yealink T48G and W52P phones are both registered and
outgoing calls work well but
incoming calls report: "You have reached a non-working number".

I set up incredible PBX 13-13.10 for Raspbian on a Raspberry PI 3 B+ (http://nerdvittles.com/?p=21255),
have managed to get it to register with both:
our SIP.US trunks and
our Yealink T48Gg and Yealink W52P phones.

I am missing how to correctly set up an incoming route which rings and connects to the phones.
I am clearly not understanding the http://192.168.1.**/admin/config.php?display=did# GUI interface
or missing required intermediate configuration steps.

 

[wardmundy]

@Milton Alvis: Place a call to your SIP.US number and watch the Asterisk CLI (asterisk -rvvvvvvvvvv) to see what happens when the call arrives. It will probably report the DID of your incoming call. In the GUI, create an Inbound Route and insert this DID as it was delivered in the DID field. Then choose a Destination for the calls, e.g. the extension of one of your Yealink phones or the Ring Group associated with all of your Yealink phones.

 

[Milton Alvis]

Thank you Ward,

My install did not have "Asterisk CLI" under the "Admin" menu.
I found the option to add "Asterisk CLI" under the "Admin" > "Module Admin" menu, installed it and now have "Asterisk CLI" as an option under the "Admin" menu.

Opening the "Asterisk CLI" option show a "CLI Command" window but I have not figured out how to implement the "watch the Asterisk CLI (asterisk -rvvvvvvvvvv)" option which you suggested.

Calling in via the SIP.US trunk, nothing shows in the
"Asterisk CLI" > "CLI Command" window and
I continue to receive the
"You have reached a non-working number"
message.

 

[geopeterwc]

@Milton Alvis ... To view the "Asterisk CLI" that Ward was referring to, use Putty or another SSH tool to log into "root" of your iPBX. Then, at the prompt, type "asterisk -rvvvvvvvvvv" (without the quotes). Then you're in the Asterisk CLI. Accessing the Asterisk CLI by way of the GUI is (my opinion) only marginally useful for this purpose. But, using SSH to log into your iPBX server and then viewing the Asterisk activity is the best way to see what's happening.
/Pete./

 

[Milton Alvis]

Thank you Pete.

I tried your clarified method.
Calling in through SIP.US
(which works if forwarded to my cell phone),
when not forwarded,
I received the same "You have reached a non-working number"
and
I did not see any responses within the Putty SSH CLI window.

 

[Milton Alvis]

@Milton Alvis: Place a call to your SIP.US number and watch the Asterisk CLI (asterisk -rvvvvvvvvvv) to see what happens when the call arrives. It will probably report the DID of your incoming call. In the GUI, create an Inbound Route and insert this DID as it was delivered in the DID field. Then choose a Destination for the calls, e.g. the extension of one of your Yealink phones or the Ring Group associated with all of your Yealink phones.

Thank you.

Signing in as root via SSH and watching for responses after imputing the Asterisk CLI (asterisk -rvvvvvvvvvv) while calling in through SIP.US
(which works if forwarded to my cell phone),
when our SIP.US number is not forwarded,
I received the same "You have reached a non-working number" recording
and
I did not see any responses within the Putty SSH CLI window.

Thus, I am clearly still not understanding, missing some basic setup/configuration issues.

(Nice GUI interface, but like any software system, until one has a more fundamental grasp of which is happening behind the scenes, it is very easy to be lost.)

Suggestions?

 

[geopeterwc]

@Milton Alvis ... Though I haven't just tried SIP.US for DID service myself (yet), and found that the "manual setup" instructions in their help files work and may be the solution for the problem you're having:

To configure inbound calling for your DIDs, be sure to include the '1' in front of the telephone number, ex: 15612322200 when adding the Inbound route!

Examine the 'Inbound Route" for your SIP.US DID in the DID Number field to be sure that it shows the complete phone number, including the "1" in front of the phone number.

More (manual) "FreePBX configuration details are here: https://support.sip.us/hc/en-us/articles/201914237-Manual-Edit-FreePBX-Configuration

/Pete./

 

[geopeterwc]

@Milton Alvis ... CONFIRMED! I used the "manual setup" instructions from SIP.US to set up a DEMO account and number, and configured the service on an iPBX 13-13 cloud system successfully. (I did not try the configuration tool that SIP.US provides.)

It's easy to confuse your assigned trunk number and the trunk password with the DID number and your account password rather than those assigned by SIP.US ... and be certain that you use the trunk number and the password that SIP.US assigned ... and to use the same trunk number and password when setting the "register string" on the "Incoming SIP Settings" panel.

(You can find the SIP.US assigned trunk number and password on their Control Panel. Click on the "Manage Your SIP Trunks and Numbers" button ... these details will be displayed. Click the "show password" link. Copy and paste the trunk number and passwords to the trunk configuration panels.)

BE SURE to configure the SIP.US trunk as a "chan_sip" trunk (not "chan_psip")

The register string should look like: [SIPUS-TrunkNumber]:[SIPUS-TrunkPassword]@gw2.sip.us (NO brackets, though!)

If you have no other DIDs configured on your system, you can use the "default" incoming route with the "DID Number" and "Caller ID Number" set to "ANY". If you are using this route, it is not necessary to specify the incoming DID number on this panel. AND set the destination for the incoming call to one or the other of your SIP phones.

Set your OUTBOUND route to select the SIP.US trunk and you should be set.

/Pete./

 

[geopeterwc]

Yeah ... I won't sign up for SIP.US services ... expensive at $24.95 a month + state and local usage taxes for ONE SIP channel, PLUS the monthly charge for the DID number. Need two or more DID numbers? More SIP channels at $24.95/month!

The DEMO was great! Slick setup, and I would've thought that the "DEMO" number that's offered would be operational for more than four! hours! And because they welcome "dialer" and "telemarketing" traffic (scammers?) is just one more reason that I am choosing to stay away from them.