[asterisk-bugs] [JIRA] (ASTERISK-21778) astobj2.c:115 INTERNAL_OBJ: user_data is NULL followed by Segmentation fault on cancelled divert
Antony Nikrooz (JIRA)
noreply at issues.asterisk.org
Fri May 10 04:09:38 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21778?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=206271#comment-206271 ]
Antony Nikrooz commented on ASTERISK-21778:
-------------------------------------------
SIP debug:
[root at lemonade asterisk]# asterisk -r
Asterisk 1.8.17.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.17.0 currently running on lemonade (pid = 5429)
Verbosity is at least 3
lemonade*CLI> core set debug 9
Core debug was 0 and is now 9
lemonade*CLI> sip set debug on
SIP Debugging enabled
lemonade*CLI> core set verbose 9
Verbosity was 3 and is now 9
== ISDN1#30: Incoming call '3456' -> '5998'
-- ISDN1#30: received Calling Party Name 83 'Ant Nikrooz'
-- Executing [5998 at isdn-in:1] Goto("CAPI/ISDN1#30/5998-0", "from-pstn,8885998,1") in new stack
-- Goto (from-pstn,8885998,1)
-- Executing [8885998 at from-pstn:1] Set("CAPI/ISDN1#30/5998-0", "__FROM_DID=8885998") in new stack
-- Executing [8885998 at from-pstn:2] Gosub("CAPI/ISDN1#30/5998-0", "app-blacklist-check,s,1()") in new stack
-- Executing [s at app-blacklist-check:1] GotoIf("CAPI/ISDN1#30/5998-0", "0?blacklisted") in new stack
-- Executing [s at app-blacklist-check:2] Set("CAPI/ISDN1#30/5998-0", "CALLED_BLACKLIST=1") in new stack
-- Executing [s at app-blacklist-check:3] Return("CAPI/ISDN1#30/5998-0", "") in new stack
-- Executing [8885998 at from-pstn:3] Set("CAPI/ISDN1#30/5998-0", "CDR(did)=8885998") in new stack
-- Executing [8885998 at from-pstn:4] ExecIf("CAPI/ISDN1#30/5998-0", "0 ?Set(CALLERID(name)=3456)") in new stack
[2013-05-10 09:24:45] WARNING[5536]: func_callerid.c:817 callerpres_read: CALLERPRES is deprecated. Use CALLERID(name-pres) or CALLERID(num-pres) instead.
-- Executing [8885998 at from-pstn:5] Set("CAPI/ISDN1#30/5998-0", "__CALLINGPRES_SV=allowed_passed_screen") in new stack
-- Executing [8885998 at from-pstn:6] Set("CAPI/ISDN1#30/5998-0", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [8885998 at from-pstn:7] Goto("CAPI/ISDN1#30/5998-0", "ext-trunk,1,1") in new stack
-- Goto (ext-trunk,1,1)
-- Executing [1 at ext-trunk:1] Set("CAPI/ISDN1#30/5998-0", "TDIAL_STRING=SIP/Lync") in new stack
-- Executing [1 at ext-trunk:2] Set("CAPI/ISDN1#30/5998-0", "DIAL_TRUNK=1") in new stack
-- Executing [1 at ext-trunk:3] Goto("CAPI/ISDN1#30/5998-0", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
-- Executing [tdial at ext-trunk:1] Set("CAPI/ISDN1#30/5998-0", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [tdial at ext-trunk:2] GotoIf("CAPI/ISDN1#30/5998-0", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
-- Executing [tdial at ext-trunk:4] ExecIf("CAPI/ISDN1#30/5998-0", "1?Set(CALLERPRES()=allowed_passed_screen)") in new stack
-- Executing [tdial at ext-trunk:5] Set("CAPI/ISDN1#30/5998-0", "DIAL_NUMBER=8885998") in new stack
-- Executing [tdial at ext-trunk:6] GosubIf("CAPI/ISDN1#30/5998-0", "1?sub-flp-1,s,1()") in new stack
-- Executing [s at sub-flp-1:1] ExecIf("CAPI/ISDN1#30/5998-0", "0?Set(TARGET_FLP_1=5998)") in new stack
-- Executing [s at sub-flp-1:2] GotoIf("CAPI/ISDN1#30/5998-0", "0?match") in new stack
-- Executing [s at sub-flp-1:3] ExecIf("CAPI/ISDN1#30/5998-0", "0?Set(TARGET_FLP_1=5998)") in new stack
-- Executing [s at sub-flp-1:4] GotoIf("CAPI/ISDN1#30/5998-0", "0?match") in new stack
-- Executing [s at sub-flp-1:5] ExecIf("CAPI/ISDN1#30/5998-0", "1?Set(TARGET_FLP_1=5998)") in new stack
-- Executing [s at sub-flp-1:6] GotoIf("CAPI/ISDN1#30/5998-0", "1?match") in new stack
-- Goto (sub-flp-1,s,8)
-- Executing [s at sub-flp-1:8] Set("CAPI/ISDN1#30/5998-0", "DIAL_NUMBER=5998") in new stack
-- Executing [s at sub-flp-1:9] Return("CAPI/ISDN1#30/5998-0", "") in new stack
-- Executing [tdial at ext-trunk:7] Set("CAPI/ISDN1#30/5998-0", "OUTNUM=5998") in new stack
-- Executing [tdial at ext-trunk:8] Dial("CAPI/ISDN1#30/5998-0", "SIP/Lync/5998,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 18652
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 193.63.48.19:5068:
INVITE sip:5998 at 193.63.48.19:5068 SIP/2.0
Via: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK340b5616;rport
Max-Forwards: 70
From: "Ant Nikrooz" <sip:3456 at 192.168.131.67>;tag=as6553c3e9
To: <sip:5998 at 193.63.48.19:5068>
Contact: <sip:3456 at 192.168.131.67:5060;transport=TCP>
Call-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(1.8.17.0)
Date: Fri, 10 May 2013 08:24:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Ant Nikrooz" <sip:3456 at 192.168.131.67>;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 2028465138 2028465138 IN IP4 192.168.131.67
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.131.67
t=0 0
m=audio 18652 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/Lync/5998
<--- SIP read from TCP:193.63.48.19:5068 --->
SIP/2.0 100 Trying
FROM: "Ant Nikrooz"<sip:3456 at 192.168.131.67>;tag=as6553c3e9
TO: <sip:5998 at 193.63.48.19:5068>
CSEQ: 102 INVITE
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
VIA: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK340b5616;rport
CONTENT-LENGTH: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from TCP:193.63.48.19:5068 --->
SIP/2.0 183 Session Progress
FROM: "Ant Nikrooz"<sip:3456 at 192.168.131.67>;tag=as6553c3e9
TO: <sip:5998 at 193.63.48.19:5068>;tag=cac0f86c9f;epid=781A35F9DD
CSEQ: 102 INVITE
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
VIA: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK340b5616;rport
CONTACT: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
CONTENT-LENGTH: 0
ALLOW: CANCEL
ALLOW: BYE
ALLOW: UPDATE
ALLOW: PRACK
SERVER: RTCC/5.0.0.0 MediationServer
<------------->
--- (13 headers 0 lines) ---
list_route: hop: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
-- SIP/Lync-00000000 is ringing
<--- SIP read from TCP:193.63.48.19:5068 --->
SIP/2.0 180 Ringing
FROM: "Ant Nikrooz"<sip:3456 at 192.168.131.67>;tag=as6553c3e9
TO: <sip:5998 at 193.63.48.19:5068>;tag=cac0f86c9f;epid=781A35F9DD
CSEQ: 102 INVITE
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
VIA: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK340b5616;rport
CONTACT: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
CONTENT-LENGTH: 0
ALLOW: CANCEL
ALLOW: BYE
ALLOW: UPDATE
ALLOW: PRACK
SERVER: RTCC/5.0.0.0 MediationServer
<------------->
--- (13 headers 0 lines) ---
list_route: hop: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
-- SIP/Lync-00000000 is ringing
<--- SIP read from TCP:193.63.48.19:5068 --->
SIP/2.0 200 OK
FROM: "Ant Nikrooz"<sip:3456 at 192.168.131.67>;tag=as6553c3e9
TO: <sip:5998 at 193.63.48.19:5068>;tag=cac0f86c9f;epid=781A35F9DD
CSEQ: 102 INVITE
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
VIA: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK340b5616;rport
CONTACT: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
CONTENT-LENGTH: 255
SUPPORTED: 100rel
CONTENT-TYPE: application/sdp
ALLOW: ACK
SERVER: RTCC/5.0.0.0 MediationServer
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 131 1 IN IP4 193.63.48.19
s=session
c=IN IP4 193.63.48.19
b=CT:1000
t=0 0
m=audio 56834 RTP/AVP 8 101
c=IN IP4 193.63.48.19
a=rtcp:56835
a=label:Audio
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 193.63.48.19:56834
list_route: hop: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19>
set_destination: Parsing <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19> for address/port to send to
set_destination: set destination to 193.63.48.19:5068
Transmitting (NAT) to 193.63.48.19:5068:
ACK sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19 SIP/2.0
Via: SIP/2.0/TCP 192.168.131.67:5060;branch=z9hG4bK089c544f;rport
Max-Forwards: 70
From: "Ant Nikrooz" <sip:3456 at 192.168.131.67>;tag=as6553c3e9
To: <sip:5998 at 193.63.48.19:5068>;tag=cac0f86c9f
Contact: <sip:3456 at 192.168.131.67:5060;transport=TCP>
Call-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(1.8.17.0)
Content-Length: 0
---
-- SIP/Lync-00000000 answered CAPI/ISDN1#30/5998-0
== ISDN1#30: Answering for 5998
<--- SIP read from TCP:193.63.48.19:5068 --->
INVITE sip:3456 at 192.168.131.67:5060;transport=TCP SIP/2.0
FROM: <sip:5998 at 193.63.48.19:5068>;epid=781A35F9DD;tag=cac0f86c9f
TO: <sip:3456 at 192.168.131.67>;tag=as6553c3e9
CSEQ: 1 INVITE
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 193.63.48.19:5068;branch=z9hG4bK9f691ad3
CONTACT: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19;ms-opaque=0c51a5674ffa5a5a>
CONTENT-LENGTH: 0
SUPPORTED: 100rel
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
<------------->
--- (12 headers 0 lines) ---
Sending to 193.63.48.19:5068 (NAT)
<--- Transmitting (NAT) to 193.63.48.19:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 193.63.48.19:5068;branch=z9hG4bK9f691ad3;received=193.63.48.19;rport=5068
From: <sip:5998 at 193.63.48.19:5068>;epid=781A35F9DD;tag=cac0f86c9f
To: <sip:3456 at 192.168.131.67>;tag=as6553c3e9
Call-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
CSeq: 1 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3456 at 192.168.131.67:5060;transport=TCP>
Content-Length: 0
<------------>
Audio is at 18652
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 193.63.48.19:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 193.63.48.19:5068;branch=z9hG4bK9f691ad3;received=193.63.48.19;rport=5068
From: <sip:5998 at 193.63.48.19:5068>;epid=781A35F9DD;tag=cac0f86c9f
To: <sip:3456 at 192.168.131.67>;tag=as6553c3e9
Call-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
CSeq: 1 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3456 at 192.168.131.67:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 2028465138 2028465139 IN IP4 192.168.131.67
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.131.67
t=0 0
m=audio 18652 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from TCP:193.63.48.19:49333 --->
INVITE sip:3436 at 192.168.131.67;user=phone SIP/2.0
FROM: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
TO: <sip:3436 at 192.168.131.67;user=phone>
CSEQ: 12499 INVITE
CALL-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d
CONTACT: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19;ms-opaque=0c51a5674ffa5a5a>
CONTENT-LENGTH: 313
REFERRED-BY: <sip:5998 at bacardi.bolton.ac.uk;user=phone>
SUPPORTED: 100rel
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
P-ASSERTED-IDENTITY: "Ant Nikrooz"<tel:3456;phone-context=enterprise>
Privacy: id
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=root 2028465138 2028465139 IN IP4 192.168.131.67
s=session
c=IN IP4 192.168.131.67
t=0 0
m=audio 18652 RTP/AVP 8 0 3 101
c=IN IP4 192.168.131.67
a=rtcp:18653
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (17 headers 15 lines) ---
Sending to 193.63.48.19:49333 (NAT)
Using INVITE request as basis request - 6cd8d158-78b2-49ac-b9dc-40e7406875dd
Found peer 'from-Lync' for '3456;phone-context=enterprise' from 193.63.48.19:49333
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.131.67:18652
Looking for 3436 in from-lync (domain 192.168.131.67)
list_route: hop: <sip:bacardi.bolton.ac.uk:5068;transport=Tcp;maddr=193.63.48.19;ms-opaque=0c51a5674ffa5a5a>
<--- Transmitting (NAT) to 193.63.48.19:49333 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d;received=193.63.48.19;rport=49333
From: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
To: <sip:3436 at 192.168.131.67;user=phone>
Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
CSeq: 12499 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3436 at 192.168.131.67:5060;transport=TCP>
Content-Length: 0
<------------>
-- Executing [3436 at from-lync:1] Goto("SIP/from-Lync-00000001", "from-pstn,7773436,1") in new stack
-- Goto (from-pstn,7773436,1)
-- Executing [7773436 at from-pstn:1] Set("SIP/from-Lync-00000001", "__FROM_DID=7773436") in new stack
-- Executing [7773436 at from-pstn:2] Gosub("SIP/from-Lync-00000001", "app-blacklist-check,s,1()") in new stack
-- Executing [s at app-blacklist-check:1] GotoIf("SIP/from-Lync-00000001", "0?blacklisted") in new stack
-- Executing [s at app-blacklist-check:2] Set("SIP/from-Lync-00000001", "CALLED_BLACKLIST=1") in new stack
-- Executing [s at app-blacklist-check:3] Return("SIP/from-Lync-00000001", "") in new stack
-- Executing [7773436 at from-pstn:3] Set("SIP/from-Lync-00000001", "CDR(did)=7773436") in new stack
-- Executing [7773436 at from-pstn:4] ExecIf("SIP/from-Lync-00000001", "0 ?Set(CALLERID(name)=3456)") in new stack
-- Executing [7773436 at from-pstn:5] Set("SIP/from-Lync-00000001", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [7773436 at from-pstn:6] Set("SIP/from-Lync-00000001", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [7773436 at from-pstn:7] Goto("SIP/from-Lync-00000001", "ext-trunk,2,1") in new stack
-- Goto (ext-trunk,2,1)
-- Executing [2 at ext-trunk:1] Set("SIP/from-Lync-00000001", "SS=$") in new stack
-- Executing [2 at ext-trunk:2] Set("SIP/from-Lync-00000001", "TDIAL_STRING=CAPI/ISDN1/${OUTNUM}") in new stack
-- Executing [2 at ext-trunk:3] Set("SIP/from-Lync-00000001", "DIAL_TRUNK=2") in new stack
-- Executing [2 at ext-trunk:4] Goto("SIP/from-Lync-00000001", "ext-trunk,tcustom,1") in new stack
-- Goto (ext-trunk,tcustom,1)
-- Executing [tcustom at ext-trunk:1] Set("SIP/from-Lync-00000001", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [tcustom at ext-trunk:2] GotoIf("SIP/from-Lync-00000001", "1?nomax") in new stack
-- Goto (ext-trunk,tcustom,4)
-- Executing [tcustom at ext-trunk:4] ExecIf("SIP/from-Lync-00000001", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
-- Executing [tcustom at ext-trunk:5] Set("SIP/from-Lync-00000001", "DIAL_NUMBER=7773436") in new stack
-- Executing [tcustom at ext-trunk:6] GosubIf("SIP/from-Lync-00000001", "1?sub-flp-2,s,1()") in new stack
-- Executing [s at sub-flp-2:1] ExecIf("SIP/from-Lync-00000001", "1?Set(TARGET_FLP_2=3436)") in new stack
-- Executing [s at sub-flp-2:2] GotoIf("SIP/from-Lync-00000001", "1?match") in new stack
-- Goto (sub-flp-2,s,10)
-- Executing [s at sub-flp-2:10] Set("SIP/from-Lync-00000001", "DIAL_NUMBER=3436") in new stack
-- Executing [s at sub-flp-2:11] Return("SIP/from-Lync-00000001", "") in new stack
-- Executing [tcustom at ext-trunk:7] Set("SIP/from-Lync-00000001", "OUTNUM=3436") in new stack
-- Executing [tcustom at ext-trunk:8] Set("SIP/from-Lync-00000001", "CALLERID(number)=3456") in new stack
-- Executing [tcustom at ext-trunk:9] Set("SIP/from-Lync-00000001", "CALLERID(name)=Ant Nikrooz") in new stack
-- Executing [tcustom at ext-trunk:10] Dial("SIP/from-Lync-00000001", "CAPI/ISDN1/3436,300,") in new stack
-- ISDN1#29: * Sending CALLED/CONNECTED NAME 80 'Ant Nikrooz'
-- Called CAPI/ISDN1/3436
-- CAPI/ISDN1#29/3436-1 is proceeding passing it to SIP/from-Lync-00000001
<--- Transmitting (NAT) to 193.63.48.19:49333 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d;received=193.63.48.19;rport=49333
From: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
To: <sip:3436 at 192.168.131.67;user=phone>
Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
CSeq: 12499 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3436 at 192.168.131.67:5060;transport=TCP>
Content-Length: 0
<------------>
-- ISDN1#29: received Called Party Name 83 'Ant DECT'
-- ast_channel_queue_connected_line_update( aG, °Q·, )
-- CAPI/ISDN1#29/3436-1 is making progress passing it to SIP/from-Lync-00000001
Audio is at 15914
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 193.63.48.19:49333 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d;received=193.63.48.19;rport=49333
From: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
To: <sip:3436 at 192.168.131.67;user=phone>;tag=as1397fb66
Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
CSeq: 12499 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3436 at 192.168.131.67:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 576859798 576859798 IN IP4 192.168.131.67
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.131.67
t=0 0
m=audio 15914 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
-- CAPI/ISDN1#29/3436-1 is ringing
<--- Transmitting (NAT) to 193.63.48.19:49333 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d;received=193.63.48.19;rport=49333
From: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
To: <sip:3436 at 192.168.131.67;user=phone>;tag=as1397fb66
Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
CSeq: 12499 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3436 at 192.168.131.67:5060;transport=TCP>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060' in 6400 ms (Method: INVITE)
== Spawn extension (ext-trunk, tdial, 8) exited non-zero on 'CAPI/ISDN1#30/5998-0'
== ISDN1#30: CAPI Hangingup for PLCI=0x201 in state 2
> ISDN1#30: CAPI INFO 0x3490: Normal call clearing
-- ISDN1#29: received Connected Party Name 83 'Ant DECT'
-- ast_channel_queue_connected_line_update( aG, °Q·, )
-- CAPI/ISDN1#29/3436-1 answered SIP/from-Lync-00000001
Audio is at 15914
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 193.63.48.19:49333 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK387a5a9d;received=193.63.48.19;rport=49333
From: "Ant Nikrooz"<sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
To: <sip:3436 at 192.168.131.67;user=phone>;tag=as1397fb66
Call-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
CSeq: 12499 INVITE
Server: FPBX-2.10.0(1.8.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3436 at 192.168.131.67:5060;transport=TCP>
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 576859798 576859799 IN IP4 192.168.131.67
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.131.67
t=0 0
m=audio 15914 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from TCP:193.63.48.19:49333 --->
ACK sip:3436 at 192.168.131.67:5060;transport=TCP SIP/2.0
FROM: <sip:3456;phone-context=enterprise at bolton.ac.uk;user=phone>;epid=781A35F9DD;tag=e73cdd2af9
TO: <sip:3436 at 192.168.131.67;user=phone>;tag=as1397fb66
CSEQ: 12499 ACK
CALL-ID: 6cd8d158-78b2-49ac-b9dc-40e7406875dd
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 193.63.48.19:49333;branch=z9hG4bK609aee71
CONTENT-LENGTH: 0
USER-AGENT: RTCC/5.0.0.0 MediationServer
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from TCP:193.63.48.19:5068 --->
ACK sip:3456 at 192.168.131.67:5060;transport=TCP SIP/2.0
FROM: <sip:5998 at 193.63.48.19:5068>;epid=781A35F9DD;tag=cac0f86c9f
TO: <sip:3456 at 192.168.131.67>;tag=as6553c3e9
CSEQ: 1 ACK
CALL-ID: 06b2493b76bd5d81779981400fdc87a9 at 192.168.131.67:5060
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 193.63.48.19:5068;branch=z9hG4bK5e942b70
CONTENT-LENGTH: 311
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
v=0
o=root 576859798 576859799 IN IP4 192.168.131.67
s=session
c=IN IP4 192.168.131.67
t=0 0
m=audio 15914 RTP/AVP 0 8 3 101
c=IN IP4 192.168.131.67
a=rtcp:15915
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (10 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.131.67:15914
[2013-05-10 09:24:53] ERROR[5502]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL
lemonade*CLI>
Disconnected from Asterisk server
[root at lemonade asterisk]# /usr/sbin/safe_asterisk: line 145: 5485 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
(Restricted to Users role)
> astobj2.c:115 INTERNAL_OBJ: user_data is NULL followed by Segmentation fault on cancelled divert
> ------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-21778
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-21778
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 1.8.17.0
> Environment: Centos 5.8 (kernel 2.6.18) with chan_capi 1.1.4 and Divas4linux 3.0 branch
> Reporter: Antony Nikrooz
> Severity: Critical
>
> This issue is reproducable in the following way:
> Call comes in on ISDN (CAPI) channel, Asterisk connects to SIP (going via MS Lync to Exchange 2010 UM). UM server tries to forward call to another extension on ISDN (Alcatel PABX). Lync sends SIP INVITE to Asterisk, ISDN line starts ringing.
> If I then hang up incoming call (ISDN->*sk->Lync), the Lync->*sk->ISDN call doesn't stop ringing. When call is answered, Asterisk crashes:
> [2013-05-10 09:24:53] ERROR[5502]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL
> lemonade*CLI>
> Disconnected from Asterisk server
> [root at lemonade asterisk]# /usr/sbin/safe_asterisk: line 145: 5485 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
> Asterisk ended with exit status 139
> Full SIP debug will be attached
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