Summary: | ASTERISK-07538: chan_sip allows INVITEs from unknown peers | ||
Reporter: | Roy Sigurd Karlsbakk (rkarlsba) | Labels: | |
Date Opened: | 2006-08-15 14:22:52 | Date Closed: | 2011-06-07 14:08:17 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | 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. |