[OpenSIPS-Users] 180 Ringing crashes OpenSIPs
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Mon Nov 3 11:18:49 CET 2008
Hi Jeff,
A fix is now available on SVN - please update and re-enable the
early_media param and see if everything ok.
Best regards,
Bogdan
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;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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