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Summary:ASTERISK-07538: chan_sip allows INVITEs from unknown peers
Reporter:Roy Sigurd Karlsbakk (rkarlsba)Labels:
Date Opened:2006-08-15 14:22:52Date Closed:2011-06-07 14:08:17
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Connecting to asterisk with an unknown sip peer, not listed in sip.conf, and no realtime enabled, asterisk disallows the REGISTER, but allows the INVITE to pass through to the default SIP context. Only one peer is registered in sip.conf, and that's not this one...

roy

****** ADDITIONAL INFORMATION ******

Asterisk SVN-trunk-r39832M, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details.
=========================================================================
Connected to Asterisk SVN-trunk-r39832M currently running on pstngwx (pid = 11740)
   -- Remote UNIX connection
Verbosity is at least 3
pstngwx*CLI> sip debug
SIP Debugging enabled
pstngwx*CLI>
<-- SIP read from 85.166.12.216:11735:
INVITE sip:22559240@Briiz SIP/2.0
Via: SIP/2.0/UDP 85.166.12.216:5060;rport;branch=z9hG4bK0FEC3B852C9311DB9BEF000D93630EAE
From: 1990068 <sip:1990068@Briiz>;tag=266606669
To: <sip:22559240@Briiz>
Contact: <sip:1990068@85.166.12.216:5060>
Call-ID: 0FD9AD71-2C93-11DB-9BEF-000D93630EAE@192.168.2.6
CSeq: 23902 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 210

v=0
o=1990068 77691254 77691348 IN IP4 85.166.12.216
s=X-Lite
c=IN IP4 85.166.12.216
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (11 headers 10 lines)---
Sending to 85.166.12.216 : 11735 (NAT)
Using INVITE request as basis request - 0FD9AD71-2C93-11DB-9BEF-000D93630EAE@192.168.2.6
Found no matching peer or user for '85.166.12.216:11735'
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 85.166.12.216:8000
Found description format pcma for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 85.166.12.216:8000
Looking for 22559240 in default (domain Briiz)
list_route: hop: <sip:1990068@85.166.12.216:5060>
Transmitting (NAT) to 85.166.12.216:11735:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.166.12.216:5060;branch=z9hG4bK0FEC3B852C9311DB9BEF000D93630EAE;received=85.166.12.216;rport=11735
From: 1990068 <sip:1990068@Briiz>;tag=266606669
To: <sip:22559240@Briiz>
Call-ID: 0FD9AD71-2C93-11DB-9BEF-000D93630EAE@192.168.2.6
CSeq: 23902 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:22559240@213.160.242.24>
Content-Length: 0
pstngwx*CLI>

---
   -- Executing [22559240@default:1] Progress("SIP/Briiz-006cd980", "") in new stack
Audio is at 213.160.242.24 port 12330
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 85.166.12.216:11735:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 85.166.12.216:5060;branch=z9hG4bK0FEC3B852C9311DB9BEF000D93630EAE;received=85.166.12.216;rport=11735
From: 1990068 <sip:1990068@Briiz>;tag=266606669
To: <sip:22559240@Briiz>;tag=as31c4d1bc
Call-ID: 0FD9AD71-2C93-11DB-9BEF-000D93630EAE@192.168.2.6
CSeq: 23902 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:22559240@213.160.242.24>
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 11740 11740 IN IP4 213.160.242.24
s=session
c=IN IP4 213.160.242.24
t=0 0
m=audio 12330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
   -- Executing [22559240@default:2] PlayTones("SIP/Briiz-006cd980", "info") in new stack
   -- Executing [22559240@default:3] Wait("SIP/Briiz-006cd980", "10") in new stack
Comments:By: Russell Bryant (russell) 2006-08-15 16:50:55

This is not a bug.  This is the default configuration.  If you would like to disable this, you set "allowguest=no" in the general section of sip.conf.  To double check how this is set, look at "sip show settings" to see the values of global SIP settings.