No route to destination when making incoming calls

This is my 1. Asterisk server and I’m having some problems that are probably very simple to solve but I’ve been staring blind at sip.conf and extensions.conf so I hope someone can help.

From my 3 snom m3 phones I can make outgoing calls
This is output from asterisk cli when making incoming calls:
– Executing [46488@musimi_incoming:1] Dial(“SIP/YYYYYYYYYY-ZZZZZZZZ”, “SIP/101| 4”) in new stack
[Jan 31 21:55:36] WARNING[2634]: app_dial.c:1202 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/YYYYYYYYYY-ZZZZZZZZ’ status is ‘CHANUNAVAIL’

At this point localnumber 102 is ringing and I have no idea why :frowning:

extension.conf:
[general]
;
static=yes
writeprotect=no
;
[globals]
;
[lokalnumre]
exten=> _XXXX,1,Dial(SIP/${EXTEN})
exten => _XXXXXXX.,1,Dial(SIP/trunk1/${EXTEN})
;
[musimi_out]
exten => _0xxxxxxxx,1,Dail(Sip/musimi/${EXTEN:1},40,r) ;indland
exten => _000x.,1,Dail(Sip/musimi/${EXTEN:1},120) ;udland
exten => _0xxxxxxxx,2,Congestion ;indland
exten => _000x.,2,Congestion ;udland
;
[alarm_112]
exten => _112,1,Dail(Sip/musimi/${EXTEN},120) ;Alarm opkald
exten => _112,2,Congestion ;indland
;
[dialout]
include => lokalnumre
include => musimi_out
include => parkedcalls
include => alarm_112
;
[musimi_incoming]
exten => 46488,1,Dial(SIP/101, 4)

sip.conf:
[general]
context=dialout
realm=domain.org ; skriv fx. dit domain her
port=5060
bindaddr=0.0.0.0
srvlookup=yes
externip=ipaddress
localnet=10.0.2.0/255.255.255.0
outboundproxy=gw1.fonet.dk
register=>YYYYYYYYYY:ZZZZZZZZ@gw1.fonet.dk/46488
disallow=all
allow=ULAW
;
[musimi]
type=friend
host=gw1.fonet.dk
dtmfmode=rfc2833
canreinvite=no
username=YYYYYYYYYYY
secret=ZZZZZZZZZZZ
context=musimi_incoming
fromuser=YYYYYYYYYY
fromdomain=gw1.fonet.dk
insecure=port,invite
nat=yes
;
[101]
type=friend
host=dynamic
username=101
secret=101 ; SSSS = kode
context=dialout
nat=no
mailbox=1101
;
[102]
type=friend
host=dynamic
username=102
secret=102 ; SSSS = kode
context=dialout
nat=no
mailbox=1102
;
[103]
type=friend
host=dynamic
username=103
secret=103 ; SSSS = kode
context=dialout
nat=no
mailbox=1103

Best I can tell is you have a context issue. You are looking for exten 101 in the [musimi_incoming] context. Yet in sip.conf you have it listed in the [dialout] context. There is no include statement between those two contexts. You need to set up a context where you can place your phones (i.e. [phones]) and then include => phones in [musimi_incoming]. There are other ways to set that up that is just what came to mind first. Good Luck.