[asterisk-users] LAN sip-to-sip

thufir hawat.thufir at gmail.com
Mon Feb 16 14:50:39 CST 2015

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a 
starfish on it.  In some ways, astonishing that it's not really that 
definitive, it's more general -- and it only clocks in at one ream of 

In any event, I'm having some port problems on my home network:


I need to open ports for Asterisk to work even on a local level.

so I'm just asking in general.  For SIP to SIP peer calling, and by that 
I just mean "ring" or "beep," some sort of ping, basically, just 
configure the two softphones to use the IP address for the Asterisk box?


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
This is free software, with components licensed under the GNU General
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           
gs102/gs102               (Unspecified) D   N             0        UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0



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