[Asterisk-Users] Parking causes crash
John Todd
jtodd at loligo.com
Tue Jun 17 15:32:14 MST 2003
I think he's talking about the same problem I'm having, which is trying to park with "#" transfers (at least, his symptoms are identical to mine.)
I sent a message with full debug on 2003-06-03 but it for some reason didn't reach the -dev list, at least according to the archives on lists.digium.com. I've enclosed it below.
JT
>Describe that a little bit.
>The call came on what interface and then you dialed what interface
>and how did you park it ? You pressed a flash button or '#' key ?
>
>Martin
>
>On Tue, 17 Jun 2003, John Congdon wrote:
>
>> Has this been solved? When I park a call, the caller hears a second of
>> music on hold
>> and then the whole system crashes.
>>
>> I can restart with a simple (asterisk -cvvv), I don't have to reboot or
>> anything
>>
> > John
--- previous message
>Date: Tue, 3 Jun 2003 12:01:01 -0700
>To: asterisk-dev at lists.digium.com
>From: John Todd <jtodd at loligo.com>
>Subject: Segfault in parking and/or SIP routines
>
>
>
>Config:
>
>SIP ATA-186 extension 2204
>SIP Cisco 7960 extension 2203
>
>Both are configured "canreinvite=no". Both phones are behind a NAT. Call parking is turned on, extension 700, completely "out of the box" config for parking. Music on hold is enabled. Further configs available upon request.
>
>*CLI> show version
>Asterisk CVS-06/02/03-20:53:53 built by root at somewhere.something.com on a i686 running Linux
>
>Symptoms:
>I call 2203 from 2204. Call is answered, progresses normally. I hit "#" and get "transfer" prompt. I type "700#" and I hear 'seven zero one'. Extension 2204 immediately hears the music-on-hold while the digits are being read. Then, I get a segmentation fault.
>
>
>
>[root at ms1 asterisk]# gdb /usr/sbin/asterisk /home/jtodd/core.1485
>GNU gdb Red Hat Linux 7.x (5.0rh-15) (MI_OUT)
>Copyright 2001 Free Software Foundation, Inc.
>GDB is free software, covered by the GNU General Public License, and you are
>welcome to change it and/or distribute copies of it under certain conditions.
>Type "show copying" to see the conditions.
>There is absolutely no warranty for GDB. Type "show warranty" for details.
>This GDB was configured as "i386-redhat-linux"...
>Core was generated by `asterisk -vvvgcd'.
>Program terminated with signal 11, Segmentation fault.
>Reading symbols from /lib/libdl.so.2...done.
>Loaded symbols for /lib/libdl.so.2
>Reading symbols from /lib/i686/libpthread.so.0...done.
>
>warning: Unable to set global thread event mask: generic error
>[New Thread 1024 (LWP 1478)]
>Error while reading shared library symbols:
>Can't attach LWP 1478: No such process
>Reading symbols from /usr/lib/libncurses.so.5...done.
>Loaded symbols for /usr/lib/libncurses.so.5
>Reading symbols from /lib/i686/libm.so.6...done.
>Loaded symbols for /lib/i686/libm.so.6
>.
>.
>[lots more lines about Loaded Symbols]
>.
>.
>Loaded symbols for /usr/lib/asterisk/modules/codec_g729b.so
>Reading symbols from /usr/lib/asterisk/modules/app_transfer.so...done.
>Loaded symbols for /usr/lib/asterisk/modules/app_transfer.so
>#0 0x08056c6f in ast_read (chan=0x8122320) at channel.c:89
>89 if (!chan->pvt->pvt) return 1;
>(gdb) bt
>#0 0x08056c6f in ast_read (chan=0x8122320) at channel.c:89
>#1 0x422024df in do_parking_thread (ignore=0x0) at res_parking.c:400
>#2 0x40033b9c in pthread_start_thread (arg=0x42a04be0) at manager.c:274
>(gdb)
>
>
>
>
>Asterisk Ready.
>*CLI> DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '5785382a4a72b2f149a4af52604f50e4 at 127.0.0.1' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '5785382a4a72b2f149a4af52604f50e4 at 127.0.0.1' of Request 103: Found
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '5785382a4a72b2f149a4af52604f50e4 at 127.0.0.1' of Request 103: Not Found
>DEBUG[7176]: File chan_sip.c, Line 3996 (handle_response): Registration successful
>DEBUG[7176]: File chan_sip.c, Line 3998 (handle_response): Cancelling timeout 3
> -- Registered SIP '2205' at 22.19.33.8 port 29313 expires 240
> -- Registered SIP '2204' at 22.19.33.8 port 29313 expires 240
> -- Registered SIP '2203' at 22.19.33.8 port 29282 expires 120
>DEBUG[7176]: File chan_sip.c, Line 3359 (check_user): Setting NAT on RTP to -1
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '1247005691 at 10.0.1.16' of Response 1: Found
>DEBUG[7176]: File chan_sip.c, Line 3359 (check_user): Setting NAT on RTP to -1
>DEBUG[7176]: File chan_sip.c, Line 2899 (build_route): build_route: Contact hop: <sip:2204 at 10.0.1.16:5060;user=phone;transport=udp>
> -- Executing NoOp("SIP/2204-6035", "") in new stack
> -- Executing Goto("SIP/2204-6035", "intern-post|2203|1") in new stack
> -- Goto (intern-post,2203,1)
> -- Executing Dial("SIP/2204-6035", "SIP/2203|30|Tt") in new stack
>DEBUG[16401]: File app_dial.c, Line 370 (dial_exec): SIMPLE DIAL (NO URL)
>DEBUG[16401]: File chan_sip.c, Line 608 (create_addr): Setting NAT on RTP to -1
> -- Called 2203
>DEBUG[7176]: File chan_sip.c, Line 503 (__sip_ack): Acked pending invite 102
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '4eb4095e5a97d4d84e612bb7613fcb58 at 22.19.33.10' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 608 (create_addr): Setting NAT on RTP to -1
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '5a236df81c434e7f47529ae91f160751 at 22.19.33.10' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 608 (create_addr): Setting NAT on RTP to -1
>DEBUG[7176]: File chan_sip.c, Line 608 (create_addr): Setting NAT on RTP to -1
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '4f222d860625f1b27fb2b22424d43041 at 22.19.33.10' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '4eb4095e5a97d4d84e612bb7613fcb58 at 22.19.33.10' of Request 102: Not Found
> -- SIP/2203-1b09 is ringing
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '7e49889f3e2101db2d8edb4a5ff1e379 at 22.19.33.10' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '4eb4095e5a97d4d84e612bb7613fcb58 at 22.19.33.10' of Request 102: Not Found
>DEBUG[7176]: File chan_sip.c, Line 2899 (build_route): build_route: Contact hop: <sip:2203 at 10.0.1.15:5060>
> -- SIP/2203-1b09 answered SIP/2204-6035
> -- Attempting native bridge of SIP/2204-6035 and SIP/2203-1b09
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '1247005691 at 10.0.1.16' of Response 2: Found
>DEBUG[17426]: File rtp.c, Line 302 (ast_rtp_read): RTP NAT: Using address 22.19.33.8:29319
>DEBUG[17426]: File chan_sip.c, Line 1228 (sip_rtp_read): Oooh, format changed to 8
>DEBUG[17426]: File rtp.c, Line 302 (ast_rtp_read): RTP NAT: Using address 22.19.33.8:29320
>DEBUG[16401]: File rtp.c, Line 838 (ast_rtp_write): Ooh, format changed from 0 to 4
>DEBUG[16401]: File rtp.c, Line 838 (ast_rtp_write): Ooh, format changed from 0 to 8
>DEBUG[16401]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
>DEBUG[16401]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 35 (#)
>DEBUG[16401]: File channel.c, Line 2114 (ast_channel_bridge): Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/2203-1b09)
>DEBUG[16401]: File channel.c, Line 2148 (ast_channel_bridge): Bridge stops bridging channels SIP/2204-6035 and SIP/2203-1b09
> -- Playing 'pbx-transfer'
>DEBUG[16401]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
>DEBUG[16401]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 55 (7)
>DEBUG[16401]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
>DEBUG[16401]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 48 (0)
>DEBUG[16401]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
>DEBUG[16401]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 48 (0)
>DEBUG[16401]: File rtp.c, Line 356 (ast_rtp_read): Sending pending DTMF
>DEBUG[16401]: File rtp.c, Line 146 (send_dtmf): Sending dtmf: 35 (#)
> -- Started music on hold, class 'default', on SIP/2204-6035
> == Parked SIP/2204-6035 on 701
>DEBUG[16401]: File rtp.c, Line 791 (ast_rtp_raw_write): Difference is 14512, ms is 1834
> -- Playing 'digits/7'
>DEBUG[5126]: File rtp.c, Line 791 (ast_rtp_raw_write): Difference is 22168, ms is 2791
> -- Playing 'digits/0'
> -- Playing 'digits/1'
> == Spawn extension (intern-post, 2203, 1) exited KEEPALIVE on 'SIP/2204-6035'
> -- Executing Macro("SIP/2204-6035", "record-cleanup") in new stack
>Expression is '1'
> -- Executing GotoIf("SIP/2204-6035", "1?5:2") in new stack
> -- Goto (macro-record-cleanup,s,5)
> -- Executing NoOp("SIP/2204-6035", "") in new stack
> -- Executing Hangup("SIP/2204-6035", "") in new stack
> == Spawn extension (macro-record-cleanup, s, 6) exited non-zero on 'SIP/2204-6035' in macro 'record-cleanup'
> == Spawn extension (intern-post, h, 1) exited non-zero on 'SIP/2204-6035'
> -- Stopped music on hold on SIP/2204-6035
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '1247005691 at 10.0.1.16' of Request 102: Found
>DEBUG[7176]: File chan_sip.c, Line 521 (__sip_ack): Stopping retransmission on '4eb4095e5a97d4d84e612bb7613fcb58 at 22.19.33.10' of Request 103: Found
>Segmentation fault
>[root at ms1 ~jtodd]# Ouch ... error while writing audio data: : Broken pipe
>
>[root at ms1 ~jtodd]#
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