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I installed Asterisk on my local Ubuntu 12.04 machine. After starting the asterisk [asterisk -c] I ran the command

netstat -nlpa | grep asterisk

and the output shown is this ->

tcp        0      0 127.0.0.1:5038          0.0.0.0:*               LISTEN      2956/asterisk   
tcp        0      0 192.168.32.181:8088     0.0.0.0:*               LISTEN      2956/asterisk   
udp        0      0 0.0.0.0:5000            0.0.0.0:*                           2956/asterisk   
udp        0      0 0.0.0.0:5060            0.0.0.0:*                           2956/asterisk   
udp        0      0 0.0.0.0:4520            0.0.0.0:*                           2956/asterisk   
udp        0      0 0.0.0.0:4569            0.0.0.0:*                           2956/asterisk   
unix  2      [ ACC ]     STREAM     LISTENING     25848    2956/asterisk       /var/run/asterisk/asterisk.ctl
unix  2      [ ]         DGRAM                    25971    2956/asterisk       

My queries are

  1. why asterisk is running on loopback address and not on 192.168.32.181:5038 ?
  2. Why SIP is not binding to any address? (just showing 0.0.0.0:5060)
  3. what file should I edit in order to work it as desire.

This is the output of command "sip show settings" on asterisk cli->

*CLI> sip show settings


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:        No
  Match Auth Username:    No
  Allow unknown access:   No
  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:             Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  SDP Session Name:       Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Legacy userfield parse: No
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  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, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
  Codec Order:            none
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            0 (Disabled)
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  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:                default
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk

Note: I am able to make a dummy call by registering a user (102) to default number 2006. But I am not able to register the same user from another machine in same LAN. It says "Fetching registration fails. 408 request time out". What could be the reason ?Is it because of the reason as in query 1.

Entry in sip.conf

[102]
type=friend
host=dynamic
secret=******
context=demo1

Entry in extensions.conf

[demo1]
exten => 2600,1,Dial(IAX2/guest@pbx.digium.com/s@default)
exten => 100,1,Dial(SIP/1060)
exten => 101,1,Dial(IAX2/guest@pbx.digium.com/s@default)
same => n,Hangup()

Output of command "sip show peers"

*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     
101                        (Unspecified)                            D   N      0        Unmonitored 
102/102                    192.168.32.181                           D   N      5071     Unmonitored 
1060/1060                  (Unspecified)                            D   N      0        Unmonitored 
1061/1061                  (Unspecified)                            D   N      0        Unmonitored 
4 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 3 offline]

Please let me know if more information is required. Thanks.

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1 Answer 1

up vote 3 down vote accepted

You ask multiple questions, I can answer some, but not all of them.

why asterisk is running on loopback address and not on 192.168.32.181:5038

As far as I can tell from a few searches port 5038 is the asterisk management interface. So I assume it binds to 127.0.0.1:5038 for security reasons. You wouldn't want that to be accessible from everywhere.

Why SIP is not binding to any address? (just showing 0.0.0.0:5060)

Actually it is binding to any address, since 0.0.0.0 is the any address. By binding to that address, the socket will receive packets send to any of the IPv4 addresses assigned to that machine.

Using the any address for UDP and not for TCP is a bit odd though. The socket API will let a TCP server know which of its addresses was used, when the server has bound to the any address. But for UDP it has to bind separate sockets to each of the IPs in order to know which of the addresses a packet was sent to. Thus it is not unusual to see a process use the any address for TCP but not for UDP. No idea why asterisk does it the other way around.

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1  
Just to chime in on this answer, you can set what Asterisk binds to for each of its various interfaces/channel drivers in their respective configuration files. AMI, for example, is configured in manager.conf - so if Asterisk is binding to 127.0.0.1:5038, it's because that config file told it to. Asterisk doesn't make random decisions about IP address binding - the system admin, via its config files, tells it what to do. –  Matt Jordan Jun 7 at 19:56

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