No route to destination when making incoming calls

Asterisk Asterisk Support
1

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

2

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.