CLI commands not available

I installed the FreePBX with the following:

Asterisk 1.6.0.5
dahdi-linux 2.1.0.4
dahdi-tools 2.1.0.2
FreePBX 2.5.1.1
Lame 3.97
Linux kernel 2.6.18-xenU-ec2-v1.0 (1000Hz timer)
Webmin 1.450

using following instructions: http://forum.voxilla.com/asterisk-support-forum/freepbx-cloud-freepbx-secured-optimized-amazon-ec2-33949.html

Now, I can log in and set up everything. However, XLITE phone is not registering. I get the following error message in Log:
failed for ‘XX.XX.XX.XX’- No matching peer found

(i tripled check the username, password and created new extensions but none of them gets registered)

I try to troubleshoot it by looking into debug.

I log into Unix and typed asterisk vvvr
sip debug

I get error " no such command ‘sim debug’

I also check many other commands but none of seems to working.

-I hope I gave info to get some help. Please let me know if there is something I need download or turn on or anything else I can try to get the extension registered.

Thanks

If the error is no matching peer found then the username and secret do not match.

You have many typo’s ‘it’s sip set debug’ you have an error ‘sim debug’.

The command is also ‘asterisk -vvv’ you don’t want the r.

I would start at the basics with a ‘sip show peer xxx’ and make sure your peer settings match.

  • Name : 100
    Secret :
    MD5Secret :
    Context : from-internal
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox : [email protected]
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 50
    Dynamic : Yes
    Callerid : “device” <100>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : Always
    ACL : Yes
    T38 pt UDPTL : No
    CanReinvite : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost :
    Addr->IP : (Unspecified) Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Transport : UDP
    Def. Username:
    SIP Options : (none)
    Codecs : 0xc (ulaw|alaw)
    Codec Order : (ulaw:20,alaw:20)
    Auto-Framing : No
    100 on REG : No
    Status : UNKNOWN
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs

This command worked, thank you. I wil try the other one’s you mentioned. Is there a link you can suggest, where I can see all the commands.

[[email protected] ~]# asterisk -vvv
Asterisk 1.6.0.5, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
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.

== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use ‘asterisk -r’ to connect.
[[email protected] ~]# sip set debug
-bash: sip: command not found

All of the sudden all the phones are registering now, i didn’t make any change. You just simply registered now.

I call call from one extention to another but hear the ring. However, Can not hear each other.

By the way: I noticed earlier SIP trunk was down as well and all of the suden SIP trunk also went up.

You have network problems. Started with one of the “without tears guides, PBX in a Flash, even the trixbox guide” it’s all well written by the same guy and FreePBX based. You need to get your network and NAT fixed.

I also made a type, it is ‘asterisk -rvvvv’ for command line with verbosity.

Asterisk is fully documented and the super secret website asterisk.org.