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Summary:ASTERISK-03668: The SIP terminal can not register asterisk server
Reporter:jerryzhi (jerryzhi)Labels:
Date Opened:2005-03-11 03:45:25.000-0600Date Closed:2011-06-07 14:04:41
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I had intalled the asterisk server and Xlit softphone can regitster the server,but my SIP terminal can not register.
The SIP debug information:
to 192.168.0.63:5060
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.63;branch=z9hG4bK2d04e712c
From: <sip:2000@192.168.0.3:5060>;tag=f7f3da88
To: <sip:2000@192.168.0.3>;tag=as7b9f6dbd
Call-ID: b171376c8792fd9daa8b1d1a42753ede@192.168.0.63
CSeq: 22 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@192.168.0.3>
Content-Length: 0


to 192.168.0.63:5060
Mar 11 17:18:42 NOTICE[5295]: chan_sip.c:8448 handle_request: Registration from '<sip:2000@192.168.0.3>' failed for '192.168.0.63'
Scheduling destruction of call 'b171376c8792fd9daa8b1d1a42753ede@192.168.0.63' in 15000 ms
no debug
SIP Debugging Disabled
*CLI> Mar 11 17:18:48 NOTICE[5295]: chan_sip.c:8448 handle_request: Registration from '<sip:2000@192.168.0.3>' failed for '192.168.0.63'


The configuration document is:
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 192.168.0.3 ; Address to bind to (all addresses on machine)
allow=all             ; Allow all codecs
context = default ; Send SIP callers that we don't know about here

[2000]
type=friend           ; This device takes and makes calls
username=2000         ; Username on device
secret=2000
auth=md5
nat=no
host=dynamic          ; This host is not on the same IP addr every time
reinvite=no
canreinvite=no
qualify=1000
disallow=all
allow=ulaw
allow=alaw
context=from-sip      ; Inbound calls from this host go here
Comments:By: Donny Kavanagh (donnyk) 2005-03-11 04:00:58.000-0600

Most likely not a bug, looks like password misconfiguration.

I assume by sip terminal you mean ATA, ensure the passwords match up, and also you have md5 set in sip.conf so be aware of that.  Read the wiki www.voip-info.org

edited on: 03-11-05 04:01
By: Matt O'Gorman (mogorman) 2005-03-11 04:01:42.000-0600

This is a configuration issue, please go to asterisk irc channel, or go to the asterisk users mailing list. Please stop posting bugs that are not bugs to the tracker
By: Matt O'Gorman (mogorman) 2005-03-11 04:03:15.000-0600

no negative karma, not ready to use that yet. but this is second time in the last few hours