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Summary:ASTERISK-04226: [PATCH] crash/core dump at INVITE
Reporter:Roy Sigurd Karlsbakk (rkarlsba)Labels:
Date Opened:2005-05-19 04:42:33Date Closed:2011-06-07 14:04:58
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-coredump-1.patch
Description:with the config below, asterisk coredumped..
the half-a-line fix is attached

---- excert from sip.conf
register => 9000:9000@213.160.242.135

[xxxx]
context=default
username=xxxx
;secret=xxxx
host=213.160.242.135
disallow=all
allow=alaw,ulaw
type=friend
nat=no
insecure=very
qualify=5000
Comments:By: Olle Johansson (oej) 2005-05-19 04:58:54

The usual question: Disclaimer on file?
By: Roy Sigurd Karlsbakk (rkarlsba) 2005-05-19 05:01:31

Please put this on hold
The attached patch doesn't fix the issue
By: Roy Sigurd Karlsbakk (rkarlsba) 2005-05-19 05:44:25

disclaimer will be faxed to oej next monday, since I'm not back at the office until then.
(To be forwarded to Digium)


By: Roy Sigurd Karlsbakk (rkarlsba) 2005-05-19 05:46:52

here's the SIP DEBUG output and below that the backtrace after the core dump. The dialplan has been reduced to the following for simplicity:

exten => 21972829,1,Dial(SIP/1002830,30)
exten => 21972829,2,Congestion
exten => 21972829,102,Busy

----------- SIP DEBUG ---------------
*CLI> sip debug ip 213.160.242.135
SIP Debugging Enabled for IP: 213.160.242.135
*CLI>

Sip read:
CANCEL sip:21972829@213.160.242.10:5060 SIP/2.0
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6887057825159
To: <sip:21972829@213.160.242.135:5060>
Call-ID: 062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c871a-6cfd29ce-3bc4
Max-Forwards: 70
Content-Length: 0


8 headers, 0 lines
Sending to 213.160.242.135 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c871a-6cfd29ce-3bc4;received=213.160.242.135;rport=5060
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6887057825159
To: <sip:21972829@213.160.242.135:5060>;tag=as5aa91b7c
Call-ID: 062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134
CSeq: 1 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


to 213.160.242.135:5060
Destroying call '062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134'


Sip read:
CANCEL sip:21972829@213.160.242.10:5060 SIP/2.0
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6887057825159
To: <sip:21972829@213.160.242.135:5060>
Call-ID: 062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c871a-6cfd29ce-3bc4
Max-Forwards: 70
Content-Length: 0


8 headers, 0 lines
Sending to 213.160.242.135 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 481 Call Leg Does Not Exist
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c871a-6cfd29ce-3bc4;received=213.160.242.135;rport=5060
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6887057825159
To: <sip:21972829@213.160.242.135:5060>;tag=as109c4bcd
Call-ID: 062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134
CSeq: 1 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


to 213.160.242.135:5060
Destroying call '062E14BA-EF63-4FF2-A045-BED8449E3E5D@81.175.33.134'


Sip read:
INVITE sip:21972829@213.160.242.10:5060 SIP/2.0
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6893156228632
To: <sip:21972829@213.160.242.135:5060>
Call-ID: 2A4352A8-1C44-45B6-A60F-15B128F35AF2@81.175.33.134
CSeq: 1 INVITE
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c8757-6cfe181d-53a5
Record-Route: <sip:21972829@213.160.242.135:5060;branch=z9hG4bK00000c2b213.160.242.135>
Contact: <sip:74797628@81.175.33.134:1000>
Max-Forwards: 70
User-Agent: SJphone/1.40.270d (SJ Labs)
Via: SIP/2.0/UDP 81.175.33.134:1000;rport=1000;branch=z9hG4bK51af21860131c9b1428c6dd800001ebd000002da
Content-Type: application/SDP
Content-Length: 340

v=0
o=- 3325488216 3325488216 IN IP4 81.175.33.134
s=SJphone
c=IN IP4 81.175.33.134
t=0 0
a=direction:passive
m=audio 49256 RTP/AVP 8 97 98 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16

13 headers, 15 lines
Using latest request as basis request
Sending to 213.160.242.135 : 5060 (NAT)
Calling find_user from around line 5548
Calling find_user(74797628) from around line 5548
Found peer '9000'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 81.175.33.134:49256
Found description format PCMA
Found description format iLBC
Found description format iLBC
Found description format PCMU
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Calling find_user from around line 1720
Calling find_user(9000) from around line 1720
Looking for 21972829 in ip24
list_route: hop: <sip:21972829@213.160.242.135:5060;branch=z9hG4bK00000c2b213.160.242.135>
list_route: hop: <sip:74797628@81.175.33.134:1000>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.160.242.135:5060;branch=z9hG4bK-428c8757-6cfe181d-53a5
Via: SIP/2.0/UDP 81.175.33.134:1000;branch=z9hG4bK51af21860131c9b1428c6dd800001ebd000002da
From: "Jorn-318"<sip:74797628@213.160.242.135:5060>;tag=6893156228632
To: <sip:21972829@213.160.242.135:5060>;tag=as66dd03e1
Call-ID: 2A4352A8-1C44-45B6-A60F-15B128F35AF2@81.175.33.134
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:21972829@213.160.242.10>
Content-Length: 0


to 213.160.242.135:5060
   -- Executing Dial("SIP/9000-f59e", "SIP/1002830|30") in new stack
Calling find_user from around line 1720
Calling find_user(1002830) from around line 1720
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault (core dumped)
test-sipgw1:/usr/src/nytelefoni/agi#


-------- BACKTRACE ------------
(gdb) bt
#0  0xb7e68e09 in strcasecmp () from /lib/tls/libc.so.6
#1  0xb74acba1 in find_user (name=0xb6556960 "1002830") at chan_sip.c:1379
#2  0xb74addea in update_user_counter (fup=0x8195488, event=3) at chan_sip.c:1722
#3  0xb74ad886 in sip_call (ast=0x819d208, dest=0xb65572f0 "1002830", timeout=0) at chan_sip.c:1618
#4  0x0805ffae in ast_call (chan=0x819d208, addr=0xb65572f0 "1002830", timeout=0) at channel.c:1975
ASTERISK-1  0xb68e81cd in dial_exec (chan=0x81902d8, data=0xb6559720) at app_dial.c:832
ASTERISK-2  0x080745e6 in pbx_exec (c=0x81902d8, app=0x814a220, data=0xb6559720, newstack=1) at pbx.c:469
ASTERISK-3  0x08076b77 in pbx_extension_helper (c=0x81902d8, context=0x8190430 "ip24", exten=0x8190524 "21972829",
   priority=1, callerid=0x8124238 "\"Jorn-318\" <74797628>", action=1) at pbx.c:1288
ASTERISK-4  0x08077bd6 in ast_spawn_extension (c=0x81902d8, context=0x8190430 "ip24", exten=0x8190524 "21972829",
   priority=1, callerid=0x8124238 "\"Jorn-318\" <74797628>") at pbx.c:1769
ASTERISK-5  0x08077ff2 in ast_pbx_run (c=0x81902d8) at pbx.c:1828
ASTERISK-6 0x08078a6f in pbx_thread (data=0x81902d8) at pbx.c:1992
ASTERISK-7 0xb7fccb63 in start_thread () from /lib/tls/libpthread.so.0
ASTERISK-8 0xb7ec718a in clone () from /lib/tls/libc.so.6
(gdb)
By: Roy Sigurd Karlsbakk (rkarlsba) 2005-05-19 06:50:28

I apologize for this, but please close this bug.
it was due to what I posted on a bug a few days ago. my code wasn't good, but was fixed before it went into cvs on stable. I had forgot to update my code... my fault :-{

sorry

roy
By: Olle Johansson (oej) 2005-05-19 08:11:41

Happens to all of us :-)

/O