[asterisk-users] SIP show peers: UNREACHABLE

[thufir]

I'm trying to configure SIP trunking.  Now, I'm referencing "Asterisk 
the definitive guide", 4th ed.  While I don't have the page handy, I was 
reading the suggestion to try SIP to SIP before proceeding to outside 
connectivity.  I'm aware that SIP trunking is a construct, but am, 
obviously, learning the system.

What I'd like to do is from the CLI "ping" either the peer below, or a 
peer somewhere.  Unfortunately, I'm also in a double+ NAT situation at 
the moment.  While Skype works (mostly) from my LAN, the connection 
isn't the greatest.  My LAN uses a wireless bridge to connect to another 
LAN.  It's just a home setup; it is what it is.

How do I test a connection?  How do  check the settings?   As far as I 
can tell, the settings are correct.


tleilax:~ #
tleilax:~ # asterisk -V
Asterisk 1.8.32.1-vici
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' 
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it 
under
certain conditions. Type 'core show license' for details.
=========================================================================
log and verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid = 
3062)
Verbosity is at least 21
tleilax*CLI>
tleilax*CLI> sip show peer babytel


   * Name       : babytel
   Secret       : <Set>
   MD5Secret    : <Not set>
   Remote Secret: <Not set>
   Context      : default
   Subscr.Cont. : <Not set>
   Language     : en
   AMA flags    : Unknown
   Netborder CPD: No
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup    :
   Pickupgroup  :
   MOH Suggest  : default
   Mailbox      :
   VM Extension : asterisk
   LastMsgsSent : 32767/65535
   Call limit   : 0
   Max forwards : 0
   Dynamic      : Yes
   Callerid     : "" <>
   MaxCallBR    : 384 kbps
   Expire       : -1
   Insecure     : no
   Force rport  : Yes
   ACL          : No
   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   : No
   Send RPID    : Yes
   TrustIDOutbnd: Legacy
   Subscriptions: Yes
   Overlap dial : No
   DTMFmode     : rfc2833
   Timer T1     : 500
   Timer B      : 32000
   ToHost       : sip.babytel.ca
   Addr->IP     : 198.38.7.11:5060
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username: 1<private>
   SIP Options  : (none)
   Codecs       : 0x4 (ulaw)
   Codec Order  : (ulaw:20)
   Auto-Framing : No
   Status       : UNREACHABLE
   Useragent    :
   Reg. Contact :
   Qualify Freq : 60000 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess     : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   Encryption   : No

tleilax*CLI>
tleilax*CLI> sip show peers
Name/username             Host Dyn Forcerport ACL Port     Status
201/201                   (Unspecified) D   N             0        UNKNOWN
babytel/1<private> 198.38.7.11                              D   
N             5060 UNREACHABLE
gs102/gs102               (Unspecified) D   N             0        UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 
offline]
tleilax*CLI>



thanks,

Thufir