Find Me / Follow Me not working

Hello guys! I have a FreePBX v 13.0.192.19 at one office and v v14.0.1.24 at another office.

Setting up- a Virtual extension with FM/FM enabled with an outside phone number in the follow me list works in v 13, but does not work with an identical configuration in v 14.0.1.24.

Anyone know of recent bugs or some change that would prevent FM/FM from working on the newer server?

When running an SIP trace, I get this error when dialing the virtual extension:

[2018-01-29 12:14:36] WARNING[23859][C-0000033a]: pbx.c:4416 __ast_pbx_run: Channel ‘SIP/201-00001357’ sent to invalid extension but no invalid handler: context,exten,priority=ext-local,return,1
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

It looks pretty straightforward to me.

In your “ext-local” context, there is not extension identified as ‘return’ or it doesn’t have a priority ‘1’.

See if you can find the “ext-local” context in each server and compare.

Sorry, but where would I find this ‘ext-local’ context?

Log in as root on the console and look in the /etc/asterisk/extension*.conf files.

ok, when I look at the extensions.conf file and grep for ext-local (in both servers) I get the same result:

; forces ext-findmefollow to take precedence over ext-local. Also exposed to
include => ext-local
include => ext-local

Good start. Now you need to find where the context is actually defined. extension*.conf (with the little star) is where you need to look. The include is good (that means the context is going to get looked at) but the definition of the context is somewhere else.

Ok - I see in the extensions_additional.conf there is this line: (304 is the virtual extension)

exten => 304,1,GotoIf($[${DB_EXISTS(AMPUSER/${EXTEN}/followme/ddial)} != 1 | “${DB(AMPUSER/${EXTEN}/followme/ddial)}” = “EXTENSION”]?ext-local,${EXTEN},1:followme-check,${EXTEN},1)

Related to this https://issues.freepbx.org/browse/FREEPBX-16645

The issue is in 13 and 14. You just probably havent upgraded 13 fully

excuse my ignorance, but does your reply mean that this is a known issue (and most likely not a result of me improperly configuring things)?

Checkout @tm1000 link he provided and you will answer your own question. :+1:t2::wink:

After installing the ‘edge’ module that was just released (in the above bug tracker link) My Follow Me issues are still unresolved. I am wondering if that bug actually had anything to do with my problem.

Debug output when calling virtual extension (304) that has an outside phone number in the follow me list. It rings once, and then gives a busy signal before ending the call.

https://pastebin.com/NEaWDDrA

In case you missed this response

I did see that response. The reason I posted again here is because I am not confident it is actually a bug as opposed to something I am configuring wrong. I had hoped to get clarification before I submit a bug ticket and waste people’s time.

I don’t think I have configured something wrong - so I suppose I will submit a bug report.

Didn’t read in detail the above posts. Did you try: 1) Create another virtual extension to see if you have the same issue? 2) In that extension advanced settings tab, change Queue State Detection to Ignore State. (Iv’e seen this making trouble some times)

  1. Yes. I have tried nearly everything including creating multiple virtual extensions, doing a side-by-side config check with the 13x server that is working properly, and using different outbound routes for the remote DID…

  2. Ignoring Queue state has still not resolved the problem.

Thanks for the suggestion. I am at a loss

The log line you included in post #1 does relate to the linked ticket, but does not support your report of FMFM not working, there is something else going on here. I have just tested by creating a virt exten and enabling FMFM to a pstn number, and all works as expected.

We are going to need a call trace:
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

I ran ‘tail’ right when I dialed the virtual extension. There may be other debug output (as this is a call center).

Please let me know if I can make it easier to read by changing verbosity or running another command to limit it. I do have SIP trace running on my phone’s IP.

https://pastebin.com/ktbzZNMR

Sorry for all the replies: Here is the result of ‘tail’ with debug turned off. In previous comments, I added a screenshot of my config as well as another pastebin link showing the results of ‘tail’ with debug turned to 2.

Tail: Debug off: https://pastebin.com/nhm81beq

Tail: Debug On: https://pastebin.com/ktbzZNMR

Your provider is blocking this call

[2018-02-07 10:08:54] DEBUG[16721][C-0000051e] channel.c: Channel 0x7f5ab4003920 'SIP/AAHTrunk-00001dfb' hanging up.  Refs: 2
[2018-02-07 10:08:54] DEBUG[16721][C-0000051e] chan_sip.c: Hangup call SIP/AAHTrunk-00001dfb, SIP callid [email protected]
[2018-02-07 10:08:54] DEBUG[16721][C-0000051e] channel.c: Channel 0x7f5ab4003920 'SIP/AAHTrunk-00001dfb' destroying
[2018-02-07 10:08:54] VERBOSE[16721][C-0000051e] app_dial.c: **Everyone is busy/congested at this time (1:0/0/1)**