Problem with inbound calls

Hi !

In recent days, incoming calls no longer work.
I don’t have problems with outgoing calls. I haven’t changed the configuration of FreePBX and I don’t understand why it is not working. Here is the configuration of my inbound route:

Description: sip_name
DID Number: sip_phone_number

Set destination:
Extensions: <123> my_extension

Nothing else.

Then here is the configuration of the trunk:

General settings:
Outbound Caller ID: sip_phone_number

Outgoing settings:
Trunk Name: sip_name
PEER Details:
type=peer
username=0033XXXXXXXXX
fromuser=0033XXXXXXXXX
secret=XXXXXXXX
host=sip.name.net
qualify=yes
insecure=very
fromdomain=name.net
context=from-trunk
allow=ulaw&g711&alaw&gsm

Incoming Settings:
USER Context: 0033XXXXXXXXX
USER Details:
type=user
secret=XXXXXXXX
context=from-trunk
qualify=no
host=dynamic

Registration:
Register String:
0033XXXXXXXXX:[email protected]

What is wrong ?
Why everything is working fine and now I can’t call the DID number ?

Thanks for the help.

When I call the DID number, FreePBX says something like that (I’m french): ‘the phone number you have dialed is not in use’…

Here is that I can see in asterisk CLI:

-- Called xxxxx3/09xxxxxxxx
-- Executing [[email protected]:1] NoOp("SIP/acK81SV26B9Q-09f35c78", "No DID or CID Match") in new stack
-- Executing [[email protected]:2] Answer("SIP/acK81SV26B9Q-09f35c78", "") in new stack
-- Executing [[email protected]:3] Wait("SIP/acK81SV26B9Q-09f35c78", "2") in new stack
-- SIP/xxxxxx3-09f31d00 answered SIP/118-09f29750
-- Packet2Packet bridging SIP/118-09f29750 and SIP/xxxxxx3-09f31d00
-- Executing [[email protected]:4] Playback("SIP/acK81SV26B9Q-09f35c78", "ss-noservice") in new stack
-- <SIP/acK81SV26B9Q-09f35c78> Playing 'ss-noservice' (language 'en')
-- Executing [[email protected]:5] SayAlpha("SIP/acK81SV26B9Q-09f35c78", "") in new stack
-- Executing [[email protected]:6] Hangup("SIP/acK81SV26B9Q-09f35c78", "") in new stack

== Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/acK81SV26B9Q-09f35c78’
– Executing [[email protected]:1] Hangup(“SIP/acK81SV26B9Q-09f35c78”, “”) in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/acK81SV26B9Q-09f35c78’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/118-09f29750’ in macro ‘dialout-trunk’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/118-09f29750’
– Executing [[email protected]:1] Macro(“SIP/118-09f29750”, “hangupcall|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/118-09f29750”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)

Thanks.

Try changing

Registration:
Register String:
0033XXXXXXXXX:[email protected]

to

0033XXXXXXXXX:[email protected]/sip_phone_number

If that fails, see How to get the DID of a SIP trunk when the provider doesn’t send it (and why some incoming SIP calls fail)

Nice !

Changing the register string with /sip_phone_number on the end of the register line is good.
Now I can receive incoming calls. It strange that this is working before that.

Thanks a lot.