AsteriskNOW IP PBX behind NAT, clients cannot connect from outside Network

Question

This is my first time working with asterisk (basically i know nothing, so bear with me)

i am running Asterisk 11.6 in a virtualbox with 512/kbps internet connection, which is behind NAT.

have two extension 1001 and 1002, these are the situations that is happening to me.

Number 1: call within local using softphone works. "no problem".

Number 2: call from outside (softphone) to local works. "no problem".

Number 3: call from local to outside, just hangs up quickly. "PROBLEM".

Number 4:call from outside to outside, never works. I can hear dial tone but no response from the receiver. "PROBLEM".

I tried forwarding port 5060 both tcp and udp nothing changes...

i also read in somewhere that i have NAT loopback error, at this point it doesnt concerns me.

My problem is i want to connect these two extensions from outside networks...

(1001)Network1--->(server)Network2--->(1002)Network3

likewise backwards... am i missing anything?

here is my sip configuration.

Global Settings:
----------------
  UDP Bindaddress:        0.0.0.0:5060
  TCP SIP Bindaddress:    Disabled
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        Off
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             FPBX-AsteriskNOW-12.0.76(11.16.0)
  SDP Session Name:       Asterisk PBX 11.16.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Trust RPID:             No
  Send RPID:              No
  Legacy userfield parse: No
  Send Diversion:         Yes
  Caller ID:              Unknown
  From: Domain:           
  Record SIP history:     Off
  Call Events:            On
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          4294967295
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10
  Localnet:               192.168.2.0/255.255.255.0

Global Signalling Settings:
---------------------------
  Codecs:                 (gsm|ulaw|alaw|g726)
  Codec Order:            ulaw:20,alaw:20,gsm:20,g726:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            30 
  RTP Hold Timeout:       300 
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Sub. min duration       60 secs
  Sub. max duration:      3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      Yes
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set> 
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:	  UDP
  Context:                from-sip-external
  Record on feature:      automon
  Record off feature:     automon
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Keepalive:              0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               
  Tone zone:              <Not set>
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   *97

----

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      
1001/1001                 1.39.63.239                              D  Yes        Yes         A  28594    UNREACHABLE                                  
1002/1002                 106.200.190.71                           D  Yes        Yes         A  47695    OK (216 ms)

This is is from my last session.

Here user 1001 is "UNREACHABLE" why? i think that is where my problem is.

help me guys...

Also i am looking for methods to connect with PSTN and GSM.

(If you guys are from India and can help me i can actually pay you, please answer with solution for above problem then i will contact for other methods)

Have you tried using a STUN server? PS I don't think you are allowed to offer payment on this site you should remove it from your question. — Paul Whelan
No that is for business contract... and please explain about STUN server? like how to config... — user4631236

Sergey S. Sergey S. · Accepted Answer · 2015-09-05T12:00:54

You have to add externip=your_public_ip in [general] section of sip.conf. Also you have to forward RTP ports range. Usually it's 10000-20000 UDP. You can see/change this range in rtp.conf.

AsteriskNOW IP PBX behind NAT, clients cannot connect from outside Network

2 Answers