Thank you. Unfortunately, custom has already been rolled back to asterisk 16.3.0 and we don’t have a coredump from the crash. We will setup a 16.15 box inhouse and attempt to replicate this.
Dan From: asterisk-dev <asterisk-dev-boun...@lists.digium.com> On Behalf Of George Joseph Sent: Thursday, November 19, 2020 2:15 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [asterisk-dev] Encountered a crash on asterisk 16.9.0 with a PJSIP SUBSCRIBE response On Thu, Nov 19, 2020 at 12:38 PM Dan Cropp <d...@amtelco.com<mailto:d...@amtelco.com>> wrote: We have a customer who was running 16.3.0 yesterday. Almost identical packets worked yesterday. We upgraded them to 16.9.0 today and the very first time it sends the SUBSCRIBE to the number/ip, the response crashes with the following backtrace. Customer required we revert back to 16.3.0 so I'm not sure I can replicate this. Any suggestions of what to do or what to try? First, I'd suggest trying 16.15 but we'd need a full coredump/backtrace to debug further. [11/19 08:22:55.815] VERBOSE[1406] res_pjsip_logger.c: <--- Transmitting SIP request (690 bytes) to UDP:z.z.z.z:5060 ---> SUBSCRIBE sip:1234567890@y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;rport;branch=z9hG4bKPj363c70fc-88e2-4d05-b2ed-d8af5bbea7a2 From: <sip:28100@x.x.x.x;user=phone>;tag=12277ed8-61f7-4b3b-a6b0-b544cc04cd64 To: "1234567890" <sip:1234567890@y.y.y.y;user=phone>;tag=16a1f36c Contact: <sip:x.x.x.x:5060> Call-ID: mailto:36b40b3bC2_jPc@y.y.y.y<mailto:36b40b3bC2_jPc@y.y.y.y> CSeq: 27685 SUBSCRIBE Route: <sip:z.z.z.z;lr;ftag=16a1f36c> Event: refer Expires: 600 Supported: 100rel, timer, replaces, norefersub Accept: message/sipfrag;version=2.0 Allow-Events: message-summary, presence, dialog, refer Max-Forwards: 70 User-Agent: Asterisk PBX 16.9.0 Content-Length: 0 [11/19 08:22:55.817] VERBOSE[1406] res_pjsip_logger.c: <--- Received SIP response (447 bytes) from UDP:z.z.z.z:5060 ---> SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bKPj363c70fc-88e2-4d05-b2ed-d8af5bbea7a2 From: <sip:28100@x.x.x.x;user=phone>;tag=12277ed8-61f7-4b3b-a6b0-b544cc04cd64 To: "1234567890" <sip:1234567890@y.y.y.y;user=phone>;tag=16a1f36c Call-ID: mailto:36b40b3bC2_jPc@y.y.y.y<mailto:36b40b3bC2_jPc@y.y.y.y> CSeq: 27685 SUBSCRIBE Allow: INVITE, ACK, BYE, CANCEL, PRACK, REFER, NOTIFY Content-Length: 0 [11/19 08:22:55.817] ERROR[1745] channel.c: FRACK!, Failed assertion bad magic number 0x0 for object 0x7ff974095ac8 (0) [11/19 08:22:55.821] ERROR[1745] : Got 22 backtrace records # 0: /usr/sbin/asterisk(__ao2_lock+0x89) [0x45c5f9] # 1: /usr/sbin/asterisk() [0x493965] # 2: /usr/sbin/asterisk(ast_queue_control_data+0x4d) [0x4987cd] # 3: /usr/lib/asterisk/modules/chan_pjsip.so(+0x7f07) [0x7ff8ecf47f07] # 4: /usr/lib/libasteriskpj.so.2(+0x746e8) [0x7ff9985616e8] # 5: /usr/lib/libasteriskpj.so.2(+0x74c88) [0x7ff998561c88] # 6: /usr/lib/libasteriskpj.so.2(+0x75f16) [0x7ff998562f16] # 7: /usr/lib/libasteriskpj.so.2(pjsip_dlg_on_tsx_state+0x5d) [0x7ff99859566d] # 8: /usr/lib/libasteriskpj.so.2(+0xa1203) [0x7ff99858e203] # 9: /usr/lib/libasteriskpj.so.2(+0xa2431) [0x7ff99858f431] #10: /usr/lib/libasteriskpj.so.2(+0xa33f4) [0x7ff9985903f4] #11: /usr/lib/libasteriskpj.so.2(pjsip_tsx_recv_msg+0x8f) [0x7ff9985929af] #12: /usr/lib/libasteriskpj.so.2(+0xa5a75) [0x7ff998592a75] #13: /usr/lib/libasteriskpj.so.2(pjsip_endpt_process_rx_data+0x157) [0x7ff998576bd7] #14: /usr/lib/asterisk/modules/res_pjsip.so(+0x2bcdc) [0x7ff8f259ccdc] #15: /usr/sbin/asterisk(ast_taskprocessor_execute+0xce) [0x599e2e] #16: /usr/sbin/asterisk() [0x5a1520] #17: /usr/sbin/asterisk(ast_taskprocessor_execute+0xce) [0x599e2e] #18: /usr/sbin/asterisk() [0x5a1cc0] #19: /usr/sbin/asterisk() [0x5a9b2c] #20: /lib/x86_64-linux-gnu/libpthread.so.0(+0x76ba) [0x7ff9968f86ba] #21: /lib/x86_64-linux-gnu/libc.so.6(clone+0x6d) [0x7ff995ed241d] Dan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- George Joseph Asterisk Software Developer direct/fax +1 256 428 6012 Check us out at www.sangoma.com<http://www.sangoma.com/> and www.asterisk.org<http://www.asterisk.org> [cid:image001.png@01D6BE7F.0D4D1D90]
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev