[SOLVED] Extension is not available to be called

[sixdev]

Good evening, I have an unexpected behavior with a freepbx 17 asterisk 21 fresh install, here is my setup:

Trunk A - voip provider added as PJSIP trunk with server ip, login and password
Trunk B - patton smartnode no authentication, added as PJSIP trunk

Single inbound route that accepts any DID/CID and route all the calls to Ring Group 61
Ring Group 61 rings for 5 seconds and if no answer it routes the call to Ring Group 62

I can receive and make calls from both the trunks

BUT HERE IS THE DISCREPANCY:

when I receive a call from Trunk B, the extension inside Ring Group 61 shows up as EXTENSION_STATE: 8 (RINGING) and then “Extension 11 is not available to be called”. At this point the call is immediately routed to Ring Group 62.
Of course the extension is not ringing, not busy, not DND. Just available and idle. Before and after the test call it funcions normally, it can receive and make calls as usual.

If I receive a call from Trunk A, the behavior is as expected: Ring Group 61 rings for 5 seconds and then the call goes to Ring Group 62.

This weird behavior is consistent even if I reverse the test calls or if I dial Trunk A repeatedly or Trunk B repeatedly.

I also tried to remove Ring Group 61 and have the call routed to Extension 11 directly, then to Ring Group 62 if no answer: same behavior.

One difference I can spot is the international prefix +39 before the CID, Trunk A appends it, Trunk B doesn’t. I don’t think it’s relevant, I don’t have any parsing rules.

How would you debug this further?

LOGS

---

Call from Trunk A

Caller ID name is ‘+39123456789’ number is ‘+39123456789’
CW Ignore is:
CF Ignore is:
CW IN_USE/BUSY is: 1
Ringgroup confirm is :
Methodology of ring is ‘ringall’
Added extension 11 to extension map
Extension 11 cf is disabled
Extension 11 do not disturb is disabled
EXTENSION_STATE: 0 (NOT_INUSE)
Extension 11 has ExtensionState: 0
Checking CW and CFB status for extension 11
Discovered PJSIP Endpoint PJSIP/11
Ended up with real PJSIP Dial string PJSIP/11/sip:[email protected]:5060
dbset CALLTRACE/11 to +39123456789
Filtered ARG3: 11
RVOL_MODE ‘’
RVOL is:
RVOLPARENT is:

---

Call from Trunk B

Caller ID name is ‘123456789’ number is ‘123456789’
CW Ignore is:
CF Ignore is:
CW IN_USE/BUSY is: 1
Ringgroup confirm is :
Methodology of ring is ‘ringall’
Added extension 11 to extension map
Extension 11 cf is disabled
Extension 11 do not disturb is disabled
EXTENSION_STATE: 8 (RINGING)
Extension 11 has ExtensionState: 8
Checking CW and CFB status for extension 11
Extension 11 is not available to be called
Extension 11 has call waiting disabled
Filtered ARG3:
RVOL_MODE ‘’
RVOL is:
RVOLPARENT is:

---

SIDE STORY:
If I shut down this freepbx 17 installation VM and I start the old freepbx version installed on a different VM, with pretty much the same configuration, Ring Groups work as expected with all the trunks. Configured as SIP trunks though, PJSIP wasn’t available back then.
Perhaps it’s something with the Patton and PJSIP?

[kgupta (Kapil Gupta)]

hey can you please try with Queue v17.0.1.11 and if still not working then raise GITHUB jira and share the asterisk cli logs for us to debug further.

thanks

[sixdev]

Hi, thanks for your reply.

If I use Queues (17.0.1.11) the extensions ring perfectly.
Queue 81 as destination for the Inbound route, Extension 11 rings for 5 seconds, that is the only member of Queue 81, and if it doens’t pick up then the Ring Group 62 rings.

The only problem with this solution is that the call is answered immediately by the pbx and a Hold Music sounds until someone picks up. I would prefer a solution where the call is not answered, but keeps ringing until someone picks up or the call is terminated.

[sixdev]

I figured this out thanks to Queues and further playing with Ring Groups.

I removed extension 11 (Cisco SPA) and I tried a different newer phone (Grandstream) as initial Ring Group, when I called Trunk B the Grandstream showed me 2 incoming calls at the same time from the same caller.
This didn’t happen when I called Trunk A, in this case the phone showed me just the 1 call.

So the problem was my patton connected with 2 ISDN cables to the two ports of the ISDN provider, the provider would send out each call to both ports, this way the patton would receive two calls (duplicated) from its two ports and forwarding both to my pbx.

When I enabled Queues the very first call of the two was answered immediately and placed in the queue, this way the second ghost call would disappear right away, resolving the RINGING state of the extention and resulting in a proper sequence of extentions ringing.