[OpenSIPS-Users] 180 Ringing crashes OpenSIPs
bogdan at voice-system.ro
Wed Nov 5 09:29:50 CET 2008
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
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).
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,
>> Jeff Pyle wrote:
>>> 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
>>> 259 if ( !(early_media && code<200 &&
>>> (gdb) bt
>>> #0 0x003e3cbf in tmcb_func (t=0xb610ef00, type=2, ps=0x184b94) at
>>> #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
>>> 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 188.8.131.52:5060 -> 184.108.40.206:5060
>>> SIP/2.0 180 Ringing.
>>> Via: SIP/2.0/UDP
>>> Via: SIP/2.0/UDP
>>> Record-Route: <sip:220.127.116.11;lr=on;ftag=as1a627d69;did=092.a565c3d2>.
>>> From: "Jeff Pyle" <sip:02511 at 18.104.22.168>;tag=as1a627d69.
>>> To: <sip:02061 at sip.fakenet.net>;tag=as70e3a685.
>>> Call-ID: 3974f19662afbc8a7f20983c6a21218a at 22.214.171.124
>>> <mailto:3974f19662afbc8a7f20983c6a21218a at 126.96.36.199>.
>>> CSeq: 103 INVITE.
>>> User-Agent: Asterisk PBX MFLD.
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>>> Supported: replaces.
>>> Contact: <sip:02061 at 188.8.131.52>.
>>> Remote-Party-ID: "Office"
>>> <sip:02061 at 184.108.40.206>;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?
>>> Users mailing list
>>> Users at lists.opensips.org
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