No inbound calls

@bruc3 wrote:

Hi,

I have been trying to setup a PBX at home with Faktortel service and cannot get it to work.

This is the versions I am running:
Debian 9
Asterisk 15.7.2
Freepbx 14

I currently setup a soft phone extension to make and receive calls.

Outbound calls are working fine. But I cannot get inbound calls to work.

I have both Inbound and Outbound route setup already. Have also forwarded all necessary ports 5060, 10000 - 20000 and 16000 - 65000 (Faktortel guide says to forward these ports) to my Asterisk server (which is behind a NAT firewall).

This is my trunk settings below.

====OUTGOING - PEER DETAILS===
username=MYUSERNAME
type=peer
sendrpid=pai
secret=MYPASSWORD
qualify=yes
nat=yes
host=sip.faktortel.net.au
fromuser=MYUSERNAME
dtmfmode=rfc2833
disallow=all
directmedia=no
context=from-trunk
canreinvite=no
allow=alaw&ulaw

==== INCOMING - USER DETAILS===
type=user
trustrpid=yes
secret=MYPASSWORD
qualify=yes
nat=yes
insecure=port,invite
host=dynamic
dtmfmode=rfc2833
disallow=all
directmedia=no&no
context=from-trunk
canreinvite=no&no
allow=alaw&ulaw

=== REGISTER STRING ===
MYUSERNAME:MYPASSWORD@sip.faktortel.net.au

If I try to call from a mobile to my number 0291234567, I hear this:
“The number you have dialed is not in service, please check the number and try again”

It seems to hit my PBX, cause I can see below generated in asterisk CLI:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7ff09000eb50 – Strict RTP learning after remote address set to: 103.26.173.71:29094
– Executing [0291234567@from-sip-external:1] NoOp(“SIP/103.26.173.71-00000004”, “Received incoming SIP connection from unknown peer to 0291234567”) in new stack
– Executing [0291234567@from-sip-external:2] Set(“SIP/103.26.173.71-00000004”, “DID=0291234567”) in new stack
– Executing [0291234567@from-sip-external:3] Goto(“SIP/103.26.173.71-00000004”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/103.26.173.71-00000004”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“SIP/103.26.173.71-00000004”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“SIP/103.26.173.71-00000004”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“SIP/103.26.173.71-00000004”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2019-05-05 09:44:45.309 AEST.
– Executing [s@from-sip-external:6] Log(“SIP/103.26.173.71-00000004”, "WARNING,“Rejecting unknown SIP connection from 103.26.173.68"”) in new stack
[2019-05-05 09:44:30] WARNING[1217][C-00000004]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from 103.26.173.68”
– Executing [s@from-sip-external:7] Answer(“SIP/103.26.173.71-00000004”, “”) in new stack
– Executing [s@from-sip-external:8] Wait(“SIP/103.26.173.71-00000004”, “2”) in new stack
– Executing [s@from-sip-external:9] Playback(“SIP/103.26.173.71-00000004”, “ss-noservice”) in new stack
– <SIP/103.26.173.71-00000004> Playing ‘ss-noservice.ulaw’ (language ‘en’)
> 0x7ff09000eb50 – Strict RTP switching to RTP target address 103.26.173.71:29094 as source
> 0x7ff09000eb50 – Strict RTP learning complete - Locking on source address 103.26.173.71:29094
– Executing [s@from-sip-external:10] PlayTones(“SIP/103.26.173.71-00000004”, “congestion”) in new stack
– Executing [s@from-sip-external:11] Congestion(“SIP/103.26.173.71-00000004”, “5”) in new stack
== Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/103.26.173.71-00000004’
– Executing [h@from-sip-external:1] Hangup(“SIP/103.26.173.71-00000004”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/103.26.173.71-00000004’

I have been trying to get this to work for days now.

Can someone please help get these Inbound calls working?

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