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Summary:ASTERISK-19305: After reciving INVITE with FROM user without phone number asterisk crashes with segfault
Reporter:Rudi (rudolf)Labels:
Date Opened:2012-02-07 10:51:34.000-0600Date Closed:2012-03-13 11:02:12
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.9.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:SIP Trunking from Alcatel OXO 8.1 to Asterisk 1.8.8.1Attachments:
Description:After receiving INVITE with FROM user without phone number asterisk crashes with segfault. Here is INVITE message which causes asterisk to crash. This error was reproducible every time until Alcatel fixed their settings. Asterisk should never crash because of this.

<--- SIP read from UDP:192.168.1.60:5060 --->
INVITE sip:331@192.168.1.51;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Supported: 100rel,from-change,timer,histinfo
User-Agent: OXO_GW_810/059.001
Session-Expires: 43200
P-Asserted-Identity: "CALLER NAME" <sip:192.168.1.60>
History-Info: <sip:331@192.168.1.51;user=phone>;index=1
To: <sip:331@192.168.1.51;user=phone>
From: "CALLER NAME" <sip:192.168.1.60>;tag=a1356ddedbe88ad9cd6fc6c73075ed83
Contact: "CALLER NAME" <sip:192.168.1.60;transport=UDP>
Content-Type: application/sdp
Call-ID: c6d41d0369633b3db64fe2933754666b@192.168.1.60
CSeq: 590668684 INVITE
Via: SIP/2.0/UDP 192.168.1.60;rport;branch=z9hG4bK4fffd7cce8a8418856817e6685537dc1
Max-Forwards: 70
Content-Length: 219

v=0
o=default 1328537655 1328537655 IN IP4 192.168.1.60
s=-
c=IN IP4 192.168.1.60
t=0 0
m=audio 32132 RTP/AVP 8 106 0
a=sendrecv
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-15
a=ptime:20
a=maxptime:90
<------------->
--- (16 headers 11 lines) ---
Sending to 192.168.1.60:5060 (NAT)
Using INVITE request as basis request - c6d41d0369633b3db64fe2933754666b@192.168.1.60
Found peer 'alcatel' for '192.168.1.60' from 192.168.1.60:5060
 == Using SIP RTP TOS bits 184
 == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 0

Comments:By: Richard Mudgett (rmudgett) 2012-02-07 11:34:59.217-0600

Thank you for your bug report. In order to move your issue forward, we require a backtrace[1] from the core file produced after the crash. Also, be sure you have DONT_OPTIMIZE enabled in menuselect within the Compiler Flags section, then:

make install

After enabling, reproduce the crash, and then execute the backtrace[1] instructions. When complete, attach that file to this issue report.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace


By: Mark Michelson (mmichelson) 2012-02-07 12:02:20.670-0600

In addition to the backtrace, I'd also suggest handing over sip.conf as well. I copied the INVITE you provided into a SIPp scenario and pointed it at Asterisk and had no crash occur. I suspect it is because of having different settings.
By: Matt Jordan (mjordan) 2012-03-13 11:02:03.397-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines