| Summary: | ASTERISK-13661: [patch] Option C in Queue() app crashes Asterisk | ||
| Reporter: | Olivier Krief (okrief) | Labels: | |
| Date Opened: | 2009-12-20 10:59:31.000-0600 | Date Closed: | 2010-02-18 15:00:56.000-0600 |
| Priority: | Critical | Regression? | No |
| Status: | Closed/Complete | Components: | Applications/app_queue |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) M16475.diff ( 1) M16475.txt ( 2) queue_crash.diff | |
| Description: | In extension.ael, this line works OK : Queue(techsupport); Adding option C crashes Asterisk as soon queue member is answering the call : Queue(techsupport,C); SIP debug traces are included in additional information section. In this section SIP/462 in the queue member answering the call and SIP/461 is the caller. I'm not sure the SIP hardphones I used for this test support SIP "Answered Elsewhere" option (I think they do support it but I didn't configure it) but this shouldn't crash Asterisk. ****** ADDITIONAL INFORMATION ****** <--- SIP read from UDP:172.16.2.105:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.2.1:5060;branch=z9hG4bK6e88f0a5;rport From: "Nahalie Doe"<sip:461@172.16.2.1>;tag=as77e1e0ec To: <sip:462@172.16.2.105:5060;user=phone>;tag=c0a80101-3f1c24 Call-ID: 4987b533394a00ab175169f442e6ce6d@172.16.2.1 CSeq: 102 INVITE Require: timer Session-Expires: 1800;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:462@172.16.2.105:5060;user=phone> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 142 v=0 o=462 4138641 4138641 IN IP4 172.16.2.105 s=- c=IN IP4 172.16.2.105 t=0 0 m=audio 41000 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=sendrecv <-------------> --- (13 headers 8 lines) --- Found RTP audio format 8 Found audio description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 172.16.2.105:41000 list_route: hop: <sip:462@172.16.2.105:5060;user=phone> set_destination: Parsing <sip:462@172.16.2.105:5060;user=phone> for address/port to send to set_destination: set destination to 172.16.2.105, port 5060 Transmitting (no NAT) to 172.16.2.105:5060: ACK sip:462@172.16.2.105:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.2.1:5060;branch=z9hG4bK6eb11b1c;rport Max-Forwards: 70 From: "Nahalie Doe" <sip:461@172.16.2.1>;tag=as77e1e0ec To: <sip:462@172.16.2.105:5060;user=phone>;tag=c0a80101-3f1c24 Contact: <sip:461@172.16.2.1> Call-ID: 4987b533394a00ab175169f442e6ce6d@172.16.2.1 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.0 Content-Length: 0 --- Scheduling destruction of SIP dialog '4987b533394a00ab175169f442e6ce6d@172.16.2.1' in 6400 ms (Method: INVITE) set_destination: Parsing <sip:462@172.16.2.105:5060;user=phone> for address/port to send to set_destination: set destination to 172.16.2.105, port 5060 Reliably Transmitting (no NAT) to 172.16.2.105:5060: BYE sip:462@172.16.2.105:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.2.1:5060;branch=z9hG4bK767de5b2;rport Max-Forwards: 70 From: "Nahalie Doe" <sip:461@172.16.2.1>;tag=as77e1e0ec To: <sip:462@172.16.2.105:5060;user=phone>;tag=c0a80101-3f1c24 Call-ID: 4987b533394a00ab175169f442e6ce6d@172.16.2.1 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.2.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- mypbc*CLI> Disconnected from Asterisk server | ||
| Comments: | By: Clod Patry (junky) 2009-12-20 14:25:48.000-0600 ive been able to reproduce this bug (see attachment) I've no clue what is the option C, can someone give me more details? But i've been able to see that we are calling ast_bridge_call() with an invalid *peer, this is because we are calling hangupcalls() and we hangup the channel. Really not sure about this fix, but it prevent segfault. By: Olivier Krief (okrief) 2009-12-20 17:26:20.000-0600 I'm not familiar with this option C, either : I met this bug while studying this option. I applied this patch this way: cd /usr/src/asterisk-1.6.2.0 wget 'http://issues.asterisk.org/file_download.php?file_id=24827&type=bug' -O - | patch -p0 ./configure make make install Unfortunately, this still craches Asterisk : CLI> Disconnected from Asterisk server By: Clod Patry (junky) 2009-12-20 22:30:50.000-0600 Are you sure you restarted * after you applied the patch (after the make install) ? By: Olivier Krief (okrief) 2009-12-21 02:19:43.000-0600 Now, I remember I didn't (shame on me). After restarting, it doesn't crash anymore. As the main reason I opened this issue is this crash, not the option C feature, it would be OK for me to either have this issue marked closed, or either have someone more experienced than me, looking at the feature itself. Thanks for all ! By: Clod Patry (junky) 2009-12-21 08:05:17.000-0600 My goal was to prevent the crash also. To be honest, i dont understand the option C, I will let Mark Michelson (the queue master) review it and give some feedback. By: Digium Subversion (svnbot) 2010-02-18 14:58:42.000-0600 Repository: asterisk Revision: 247736 U trunk/apps/app_queue.c ------------------------------------------------------------------------ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue ASTERISK-13661) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=247736 By: Digium Subversion (svnbot) 2010-02-18 14:59:38.000-0600 Repository: asterisk Revision: 247737 _U branches/1.6.2/ U branches/1.6.2/apps/app_queue.c ------------------------------------------------------------------------ r247737 | dvossel | 2010-02-18 14:59:38 -0600 (Thu, 18 Feb 2010) | 14 lines Merged revisions 247736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue ASTERISK-13661) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=247737 By: Digium Subversion (svnbot) 2010-02-18 15:00:38.000-0600 Repository: asterisk Revision: 247738 _U branches/1.6.1/ ------------------------------------------------------------------------ r247738 | dvossel | 2010-02-18 15:00:38 -0600 (Thu, 18 Feb 2010) | 13 lines Blocked revisions 247736 via svnmerge ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue ASTERISK-13661) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=247738 By: Digium Subversion (svnbot) 2010-02-18 15:00:55.000-0600 Repository: asterisk Revision: 247739 _U branches/1.6.0/ ------------------------------------------------------------------------ r247739 | dvossel | 2010-02-18 15:00:55 -0600 (Thu, 18 Feb 2010) | 13 lines Blocked revisions 247736 via svnmerge ........ r247736 | dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines fixes Queue with C option crash (closes issue ASTERISK-13661) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=247739 | ||