Route DID direct to extension/No Route Error

[DrTooth&TheElectricMayhem]

v18 in production, testing a v20 replacement and still getting used to changes.
SRTP/TLS DID trunk establishes fine, firewall SNAT/DNAT works.

Route DID direct to T54W extension via user but I'm getting a No Route Error.
Tried making a ring group just to test.. and that doesn't work either.


I've reviewed documentation and forums but it's gotta be something i'm not seeing.

 

[SweetAction]

Review the invite (verbose mode, make a call) and make sure that your DID matches what is in the invite. Likely needs to be in E164 format. You could also try a * in front of the DID to match anything ending in the DID.

 

[DrTooth&TheElectricMayhem]

Everything is E164 AFAIK. SIP verbose doesn't show anything of note... the call negotiates fine... just routes to an end call.
I see the call in the 3CX call log as routing to a department that has no incoming rules instead of routing direct to the extension. The route to user by assigning the DID (number/trunk) to the user doesn't appear to do anything.

I made rules in the department and the phone call does go through ( to the department) now.. but it doesn't resolve my problem.

How, in v20, can i route a DID direct to an extension/phone without creating a department for each direct to extension route?

 

[SweetAction]

Does the user have a main department set? Can you remove the DID from the department, save, then assign to the user, save, and see if this issue is resolved?

 

[DrTooth&TheElectricMayhem]

Yes and Yes. It reverts back to the original error when I assign to dept, save, remove, save, assign to user, save.

 

[SweetAction]

I'm out of steps here unless you want to delete and recreate the SIP trunk (that would be my next step). It does work in my test environment and with v20 not being a stable release I can't say for sure what support would tell you.

 

[DrTooth&TheElectricMayhem]

I made each DID it's own department and that works. Bro, i'm beyond frustrated.
Now I'm having an error where the calls go right to VM instantly (with a no answer, forwarded to VM error) and they drop instantly after pickup post ring on any external calls.
I've made sure everything is E164 and the same default codecs.