[OpenSIPS-Users] 180 Ringing crashes OpenSIPs

[Jeff Pyle]

Hello,
 
We've got a handful of Asterisk boxes that register to today's build of
opensips_1_4.  All works well.  But, when we call from any of these
Asterisk boxes to one particular one, OpenSIPs crashes.  Sometimes it
relays the 180 Ringing just before crash, sometimes it crashes first.
 
Here's the backtrace:
 
Program received signal SIGSEGV, Segmentation fault.
0x003e3cbf in tmcb_func (t=0xb610ef00, type=2, ps=0x184b94) at
acc_logic.c:259
259  if ( !(early_media && code<200 &&
(gdb) bt
#0  0x003e3cbf in tmcb_func (t=0xb610ef00, type=2, ps=0x184b94) at
acc_logic.c:259
#1  0x0015c057 in run_trans_callbacks (type=2, trans=0xb610ef00,
req=0xb610fea8, rpl=0x81cff58, code=180) at t_hooks.c:205
#2  0x0016653c in t_reply_matching (p_msg=0x81cff58,
p_branch=0xbfc737f4) at t_lookup.c:840
#3  0x001669dc in t_check (p_msg=0x81cff58, param_branch=0xbfc737f4) at
t_lookup.c:911
#4  0x00177136 in reply_received (p_msg=0x81cff58) at t_reply.c:1288
#5  0x080651ca in forward_reply (msg=0x81cff58) at forward.c:507
#6  0x08095536 in receive_msg (
    buf=0x817a0a0 "SIP/2.0 180 Ringing\r\nVia: SIP/2.0/UDP
60.70.82.45;branch=z9hG4bK9027.cfa92ba.0;received=60.70.82.45\r\nVia:
SIP/2.0/UDP
208.157.201.66:5060;received=208.157.201.66;branch=z9hG4bK3206a4aa;rport
=5060\r\nRecor"..., len=697, rcv_info=0xbfc73924) at receive.c:203
#7  0x080d7ef7 in udp_rcv_loop () at udp_server.c:449
#8  0x0806d94e in main (argc=1, argv=0xbfc73b14) at main.c:780
 
Here's a packet that made it crash.  Not the time that I got this
particular backtrace, but it crashed nonetheless:
 
U +0.008071 208.157.208.67:5060 -> 60.70.82.45:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP
60.70.82.45;branch=z9hG4bK28b3.c9a41341.0;received=60.70.82.45.
Via: SIP/2.0/UDP
208.157.201.66:5060;received=208.157.201.66;branch=z9hG4bK5de91597;rport
=5060.
Record-Route: <sip:60.70.82.45;lr=on;ftag=as1a627d69;did=092.a565c3d2>.
From: "Jeff Pyle" <sip:02511 at 208.157.201.66>;tag=as1a627d69.
To: <sip:02061 at sip.fakenet.net>;tag=as70e3a685.
Call-ID: 3974f19662afbc8a7f20983c6a21218a at 208.157.201.66.
CSeq: 103 INVITE.
User-Agent: Asterisk PBX MFLD.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Contact: <sip:02061 at 208.157.208.67>.
Remote-Party-ID: "Office"
<sip:02061 at 208.157.201.66>;party=called;privacy=off;screen=no.
 
This same configuration of Asterisk boxes works fine on OpenSER 1.3.2.
Still in the process of migration...
 
Any thoughts?
 
 
Thanks,
Jeff
 
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