Inbound route issue - the number you have dialed is not in service

FreePBX Providers
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1

Hi

My FreePBX setup is working for outgoing calls but not for incoming calls. I have an inbound route connected correctly with DID=__ TARGET_PHONE_NUMBER__ which connects to the right extension, but when I try to cal, it says “the number you have dialed is not in service”. Is there a problem with the firewall or seomthing ?

[2020-02-19 16:55:52] VERBOSE[1570][C-000000b6] netsock2.c: Using SIP RTP TOS bits 184
[2020-02-19 16:55:52] VERBOSE[1570][C-000000b6] netsock2.c: Using SIP RTP CoS mark 5
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [__TARGET_PHONE_NUMBER__@from-sip-external:1] NoOp("SIP/46.31.231.185-000000ae", "Received incoming SIP connection from unknown peer to __TARGET_PHONE_NUMBER__") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [__TARGET_PHONE_NUMBER__@from-sip-external:2] Set("SIP/46.31.231.185-000000ae", "DID=__TARGET_PHONE_NUMBER__") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [__TARGET_PHONE_NUMBER__@from-sip-external:3] Goto("SIP/46.31.231.185-000000ae", "s,1") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx_builtins.c: Goto (from-sip-external,s,1)
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:1] GotoIf("SIP/46.31.231.185-000000ae", "1?setlanguage:checkanon") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx_builtins.c: Goto (from-sip-external,s,2)
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:2] Set("SIP/46.31.231.185-000000ae", "CHANNEL(language)=en") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:3] GotoIf("SIP/46.31.231.185-000000ae", "1?noanonymous") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx_builtins.c: Goto (from-sip-external,s,5)
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:5] Set("SIP/46.31.231.185-000000ae", "TIMEOUT(absolute)=15") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] func_timeout.c: Channel will hangup at 2020-02-19 16:56:07.246 UTC.
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:6] Set("SIP/46.31.231.185-000000ae", "receveip=recvip") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:7] Log("SIP/46.31.231.185-000000ae", "WARNING,"Rejecting unknown SIP connection from 46.31.231.185"") in new stack
[2020-02-19 16:55:52] WARNING[23441][C-000000b6] Ext. s: "Rejecting unknown SIP connection from 46.31.231.185"
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:8] Answer("SIP/46.31.231.185-000000ae", "") in new stack
[2020-02-19 16:55:52] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:9] Wait("SIP/46.31.231.185-000000ae", "2") in new stack
[2020-02-19 16:55:54] VERBOSE[23441][C-000000b6] pbx.c: Executing [s@from-sip-external:10] Playback("SIP/46.31.231.185-000000ae", "ss-noservice") in new stack
[2020-02-19 16:55:54] VERBOSE[23441][C-000000b6] file.c: <SIP/46.31.231.185-000000ae> Playing 'ss-noservice.alaw' (language 'en')
[2020-02-19 16:55:58] VERBOSE[23441][C-000000b6] pbx.c: Executing [h@from-sip-external:1] Hangup("SIP/46.31.231.185-000000ae", "") in new stack
[2020-02-19 16:55:58] VERBOSE[23441][C-000000b6] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/46.31.231.185-000000ae'
2

"WARNING,“Rejecting unknown SIP connection from 46.31.231.185"”

(Presumably your trunk to VoipPhone, it need to be set up correctly)

3

I don’t see any option in the Trunk to specify an IP address?

4

You would better show what you do see :slight_smile: you cant have a working trunk without it knowing what the host is.

5

Providers IPs all of them need to be listed in firewall and on the incoming trunks. If using PJ you can use the permit ips or chan sip you have to have a trunk for each incoming ip address the provider uses. Some have up to 16.

6

Some have “over” 16. There’s one (VOIP.MS, I think, maybe) that uses an entire Class-C (CIDR/24) for their source servers.

7

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