No incoming calls "number out of service" help needed

Hi,

Im having some real issues with receiving a call (no outbound problems). When i call the number i get “The number you have dailed is not in service”. I did a lot of resreach on the web and this forum, but i cant get it to work.

The voip provider is sip.sipnl.net and it uses round robin to register SIP trunks. There is no possibility to register the trunk on 1 IP, only a DNS name.

When calling to 31356932915 it also uses round robin to connect the call to the PBX.

Allow Anonymous Sip Call is ON
DID is ANY

sip_general_custom.conf
Allowguest=yes
Alwaysauthreject=no

TRUNK SETTINGS:
Trunk Name: SIPNL
Outbound CallerID: 31356932915
CID Options: ANY
MAX Channels: blank

Trunk Name: SIPNL
PEER Details:

host=sip.sipnl.net
port=5060
disallow=all
allow=ulaw
type=peer
qualify=yes
context=from-pstn
nat=no
canreinvite=no
username=31356932915
secret=passwd

User Context: empty
User Details: empty

Register String:
31356932915:[email protected]/31356932915

Inbound Route

DID: ANY
CID: ANY
Destination: extension 1000 (1000 can make outbound calls)

Extension 1000
Context= from-internal
Use NAT= yes

/var/log/asterisk/cdr-csv/Master.csv

“”,“063830****”,“s”,“from-sip-external”,“063830****”,“SIP/91.195.161.2-00000041”,"",“Playback”,“ss-noservice”,“2016-07-06 09:06:48”,“2016-07-06 09:06:48”,“2016-07-06 09:06:54”,6,6,“ANSWERED”,“DOCUMENTATION”,“1467796008.65”,""

SIP SHOW PEER 1000

  • Name : 1000
    Description :
    Secret :
    MD5Secret :
    Remote Secret:
    Context : from-internal
    Record On feature : automon
    Record Off feature : automon
    Subscr.Cont. :
    Language :
    Tonezone :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Named Callgr :
    Nam. Pickupgr:
    MOH Suggest :
    Mailbox : [email protected]
    VM Extension : 97
    LastMsgsSent : 0/0
    Call limit : 2147483647
    Max forwards : 0
    Dynamic : Yes
    Callerid : “1000” <1000>
    MaxCallBR : 384 kbps
    Expire : 1620
    Insecure : no
    Force rport : Yes
    Symmetric RTP: Yes
    ACL : Yes
    DirectMedACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: 4294967295
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : Yes
    Send RPID : No
    TrustIDOutbnd: Legacy
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : 91.213.
    **.***:10152
    Defaddr->IP : (null)
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: 1000
    SIP Options : (none)
    Codecs : (gsm|ulaw|alaw)
    Codec Order : (alaw:20,gsm:20,ulaw:20)
    Auto-Framing : No
    Status : OK (8 ms)
    Useragent : X-Lite release 4.9.0 stamp 78104
    Reg. Contact : sip:[email protected]***.***:10152;rinstance=ae008a1a7aa06850
    Qualify Freq : 60000 ms
    Keepalive : 0 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    RTP Engine : asterisk
    Parkinglot :
    Use Reason : No
    Encryption : No

sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.sipnl.net:5060 N 31356932915 105 Registered Wed, 06 Jul 2016 11:46:58


sip show users
1000 passwd from-internal Yes Yes


When more info needed, just ask and ill provide.

Thnx so much in advance

try using the DID=31356932915

Thnx for the reply Jerson

I called the provider and they set up a proxy so that the calls are always forwarded to the IP where the number is registered on. It works perfectly now :smiley: