[asterisk-users] SIP show peers: UNREACHABLE

thufir hawat.thufir at gmail.com
Sun Feb 15 19:57:40 CST 2015

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
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk, 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 
certain conditions. Type 'core show license' for details.
log and verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk currently running on tleilax (pid = 
Verbosity is at least 21
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     :
   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> sip show peers
Name/username             Host Dyn Forcerport ACL Port     Status
201/201                   (Unspecified) D   N             0        UNKNOWN
babytel/1<private>                              D   
N             5060 UNREACHABLE
gs102/gs102               (Unspecified) D   N             0        UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 



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