[OpenSIPS-Users] 180 Ringing crashes OpenSIPs

Bogdan-Andrei Iancu bogdan at voice-system.ro
Thu Nov 6 14:32:47 CET 2008


Jeff,

Any way to get access to the corefile and binaries for inspection?

Regards,
Bogdan

Jeff Pyle wrote:
> I did a fresh update of opensips-1.4 SVN, and a recompile at about 15:00
> UTC today.  It still blows up with early media.
>
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org
> [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei
> Iancu
> Sent: Wednesday, November 05, 2008 3:30 AM
> To: Richard Revels
> Cc: users at lists.opensips.org
> Subject: Re: [OpenSIPS-Users] 180 Ringing crashes OpenSIPs
>
> Hi Richard,
>
> Thanks for the update - the backtrace you get shows an older code (it
> was fixed two days ago)
>
> 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 &&
>
>
> That line looks now like:
>      if ( code<200 && !(early_media &&
>
> Please be sure you have the latest 1.4.2 (SVN 1.4 branch).
>
> Regarards,
> Bogdan
>
> Richard Revels wrote:
>   
>> I'm getting the same backtrace from core dumps I'm having.  Updating 
>> from svn didn't help but turning off the early media accounting seems 
>> to be keeping it from happening.
>>
>> Richard Revels
>>
>>
>> On Oct 30, 2008, at 11:17 AM, Bogdan-Andrei Iancu wrote:
>>
>>     
>>> 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;r
>>>> port=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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>>>>
>>>>         
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>>>       
>>     
>
>
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