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[hawat.thufir at gmail.com]

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@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

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