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<DIV><FONT face=Tahoma size=2>Hello,</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Tahoma size=2>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.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Tahoma size=2>Here's the backtrace:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Tahoma size=2>Program received signal SIGSEGV, Segmentation
fault.<BR>0x003e3cbf in tmcb_func (t=0xb610ef00, type=2, ps=0x184b94) at
acc_logic.c:259<BR>259 if ( !(early_media && code<200
&&<BR>(gdb) bt<BR>#0 0x003e3cbf in tmcb_func (t=0xb610ef00,
type=2, ps=0x184b94) at acc_logic.c:259<BR>#1 0x0015c057 in
run_trans_callbacks (type=2, trans=0xb610ef00, req=0xb610fea8, rpl=0x81cff58,
code=180) at t_hooks.c:205<BR>#2 0x0016653c in t_reply_matching
(p_msg=0x81cff58, p_branch=0xbfc737f4) at t_lookup.c:840<BR>#3 0x001669dc
in t_check (p_msg=0x81cff58, param_branch=0xbfc737f4) at
t_lookup.c:911<BR>#4 0x00177136 in reply_received (p_msg=0x81cff58) at
t_reply.c:1288<BR>#5 0x080651ca in forward_reply (msg=0x81cff58) at
forward.c:507<BR>#6 0x08095536 in receive_msg (<BR>
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<BR>#7 0x080d7ef7 in
udp_rcv_loop () at udp_server.c:449<BR>#8 0x0806d94e in main (argc=1,
argv=0xbfc73b14) at main.c:780</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Tahoma size=2>Here's a packet that made it crash. Not the
time that I got this <SPAN class=322522020-29102008>particular
</SPAN>backtrace, but it crashed nonetheless:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Tahoma size=2>U +0.008071 208.157.208.67:5060 ->
60.70.82.45:5060<BR>SIP/2.0 180 Ringing.<BR>Via: SIP/2.0/UDP
60.70.82.45;branch=z9hG4bK28b3.c9a41341.0;received=60.70.82.45.<BR>Via:
SIP/2.0/UDP
208.157.201.66:5060;received=208.157.201.66;branch=z9hG4bK5de91597;rport=5060.<BR>Record-Route:
<sip:60.70.82.45;lr=on;ftag=as1a627d69;did=092.a565c3d2>.<BR>From: "Jeff
Pyle" <sip:02511@208.157.201.66>;tag=as1a627d69.<BR>To:
<sip:02061@sip.fakenet.net>;tag=as70e3a685.<BR>Call-ID: <A
href="mailto:3974f19662afbc8a7f20983c6a21218a@208.157.201.66">3974f19662afbc8a7f20983c6a21218a@208.157.201.66</A>.<BR>CSeq:
103 INVITE.<BR>User-Agent: Asterisk PBX MFLD.<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.<BR>Supported: replaces.<BR>Contact:
<sip:02061@208.157.208.67>.<BR>Remote-Party-ID: "Office"
<sip:02061@208.157.201.66>;party=called;privacy=off;screen=no.</FONT></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><SPAN class=322522020-29102008><FONT face=Tahoma size=2>This same
configuration of Asterisk boxes works fine on OpenSER 1.3.2. Still in the
process of migration...</FONT></SPAN></DIV>
<DIV><FONT face=Tahoma size=2></FONT> </DIV>
<DIV><FONT face=Tahoma size=2><SPAN class=322522020-29102008>Any
thoughts?</SPAN></FONT></DIV>
<DIV><FONT face=Tahoma size=2><SPAN
class=322522020-29102008></SPAN></FONT> </DIV>
<DIV><FONT face=Tahoma size=2><SPAN
class=322522020-29102008></SPAN></FONT> </DIV>
<DIV><FONT face=Tahoma size=2><SPAN
class=322522020-29102008>Thanks,</SPAN></FONT></DIV>
<DIV><FONT face=Tahoma size=2><SPAN
class=322522020-29102008>Jeff</SPAN></FONT></DIV>
<DIV><FONT face=Tahoma size=2><SPAN
class=322522020-29102008></SPAN></FONT> </DIV></BODY></HTML>