[OpenSIPS-Users] 180 Ringing crashes OpenSIPs

[Bogdan-Andrei Iancu]

Hi Jeff,

It might be related to a fix I did in the ACC module for early_media - 
could you disable early_media accounting to see if it still crashes ?

Thanks and regards,
Bogdan

Jeff Pyle wrote:
> 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 
> <mailto: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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