[OpenSIPS-Users] 180 Ringing crashes OpenSIPs
rrevels at bandwidth.com
Tue Nov 4 19:16:45 CET 2008
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.
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
>> 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 22.214.171.124:5060 -> 126.96.36.199:5060
>> SIP/2.0 180 Ringing.
>> Via: SIP/2.0/UDP
>> Via: SIP/2.0/UDP
>> Record-Route: <sip:
>> From: "Jeff Pyle" <sip:02511 at 188.8.131.52>;tag=as1a627d69.
>> To: <sip:02061 at sip.fakenet.net>;tag=as70e3a685.
>> Call-ID: 3974f19662afbc8a7f20983c6a21218a at 184.108.40.206
>> <mailto:3974f19662afbc8a7f20983c6a21218a at 220.127.116.11>.
>> CSeq: 103 INVITE.
>> User-Agent: Asterisk PBX MFLD.
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
>> Supported: replaces.
>> Contact: <sip:02061 at 18.104.22.168>.
>> Remote-Party-ID: "Office"
>> <sip:02061 at 22.214.171.124>;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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