Home » Asterisk Users » SIP Show Peers: UNREACHABLE
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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 tleilax:~ #
tleilax:~ # asterisk -rm Asterisk, Copyright (C) 1999 – 2013 Digium, Inc. and others. Created by Mark Spencer
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 currently running on tleilax (pid 3062)
Verbosity is at least 21
tleilax*CLI> sip show peer babytel

* Name : babytel
Secret :
MD5Secret :
Remote Secret:
Context : default
Subscr.Cont. :
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 :
Addr->IP :
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1 SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
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 D
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0



One thought on - SIP Show Peers: UNREACHABLE

  • Page 176 of Asterisk, the definitive manual, discusses “Connecting an Asterisk system to a SIP provider” in the context of, at least the concept of, “trunking”.

    Previously, I wasn’t able to connect to the peer, and attributed that to a combination of double NAT (plus), and latency and lag due to wi-fi.
    Now that I’m directly connected to the cable modem (well, gateway router and modem combo), the connection is better and I’m able to make outgoing VoIP calls with Jitsi.

    Am I right in thinking that the very same connection parameters I
    entered in Jitsi will work fine when entered in Asterisk with syntax like:

    register => username:password@your.provider.tld

    and by creating the peer entry in sip.conf for the service provider.

    One concern is that the provider uses:

    1. User ID can be any one of your 11-digit babyTEL telephone numbers.
    Typically your main number but can be any one of your other phone
    2. For your protection the SIP Password field does not reveal your
    password until you explicitly click on ‘Show password’.
    3. If Outbound Proxy is not supported on your system, try one of the
    following two options:
    1. Add the line “” to your system’s
    “hosts” file and configure the SIP Proxy as:
    “”. This uses the TCP/IP “hosts” file address
    mapping mechanism to redirect SIP traffic to the Outbound Proxy.
    2. Configure the SIP Proxy as: “”. This replaces
    the SIP Proxy address with a resolved Outbound Proxy address.

    On a mac, I added that line to the hosts file — but I’m not sure it’s required. How do I specify the SIP proxy and the outbound proxy?
    What’s the distinction between a SIP proxy and outbound proxy?

    In Jitsi, I configured as for SIP id.

    In “Connection” I used “” for the registrar and the user,
    1234567890, as the the authorization name. I put the proxy as, port 5065 and the preferred transport as UDP. I don’t see all those options, particularly surrounding the proxy and outbound proxy. Again, I’m unclear on why there’s a proxy specificed, and then a different outbound proxy is specified as well.

    How do I establish that I’ve entered the parameters correctly in Asterisk? Or, how do I establish that the parameters are incorrectly entered? Because Jitsi is able to call out and in, I believe that eliminates NAT, firewall or other networking issues.