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Summary:ASTERISK-00785: asterisk segfaults when doing outgoing or incoming call
Reporter:digger_one (digger_one)Labels:
Date Opened:2004-01-11 17:20:11.000-0600Date Closed:2011-06-07 14:05:00
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) core.13021.gz
Description:setup:
Mandrake 9.2, custom kernel 2.4.21
libpri and zaptel cvs from 2004-01-11, using ztdummy+usb-uhci
Asterisk CVS-01/11/04-18:37:51
openh323_1.12.2
pwlib_1.5.2
ixj-3.1.0-src.tar.gz
asterisk-oh323-0.5.7.tar.gz
Internet PhoneJack PCI, driver v3.1.0
Cisco AS5300, IOS 12.2(2)XB5
X-Lite Softphone 2.0 build 1101

Problem: asterisk segfaults when doing outgoing or incoming call

test1: x-lite/sip/g711u -> asterisk -> as5300/oh323/g711a
sip.conf: disallow=all
         allow=ulaw

test2: x-lite/sip/g711a -> asterisk -> as5300/oh323/g711a
sip.conf: disallow=all
         allow=alaw


test1 is fine. But test2 (changing to g711a from x-lite to asterisk) crashes *. The same also happens if codec is changed on x-lite only (get No compatible codecs!).

However, this bug only appears if I do a Dial() to the Oh323 channel. If I connect locally to * (demo application,
echo-test, iax to digium) there is no problem.

Note that it happens when calls are initiated in both
directions.


****** ADDITIONAL INFORMATION ******

x-lite log, test1:
SEND >> 193.220.101.242:5060
INVITE sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6925006BC2F44A7F8F6C47FDDAB95D18
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2983 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 200

v=0
o=tommy 1174334283 1174334283 IN IP4 192.168.10.11
s=X-Lite
c=IN IP4 192.168.10.11
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE << 193.220.101.242:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6925006BC2F44A7F8F6C47FDDAB95D18
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as69480087
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2983 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Proxy-Authenticate: Digest realm="asterisk", nonce="776a77c8"
Content-Length: 0


SEND >> 193.220.101.242:5060
ACK sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6925006BC2F44A7F8F6C47FDDAB95D18
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as69480087
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2983 ACK
Max-Forwards: 70
Content-Length: 0


SEND >> 193.220.101.242:5060
INVITE sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK1E4EC6B1AF90475AB7BA1998D8BD0B14
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2984 INVITE
Proxy-Authorization: Digest username="tommy",realm="asterisk",nonce="776a77c8",response="a10b3424e176c703027588efd2e8a49e",uri="sip:095128304@home"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 200

v=0
o=tommy 1174334303 1174334303 IN IP4 192.168.10.11
s=X-Lite
c=IN IP4 192.168.10.11
t=0 0
m=audio 8000 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE << 193.220.101.242:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK1E4EC6B1AF90475AB7BA1998D8BD0B14
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as45c544eb
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2984 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Content-Length: 0


RECEIVE << 193.220.101.242:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK1E4EC6B1AF90475AB7BA1998D8BD0B14
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as45c544eb
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2984 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Content-Type: application/sdp
Content-Length: 195

v=0
o=root 12959 12959 IN IP4 193.220.101.242
s=session
c=IN IP4 193.220.101.242
t=0 0
m=audio 15186 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

SEND >> 193.220.101.242:5060
ACK sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK21DC7E111DD341F18A0E501DBBE5687A
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2984 ACK
Max-Forwards: 70
Content-Length: 0


SEND >> 193.220.101.242:5060
BYE sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKFCF7BB8465974CFAA116F2BE6F28F50C
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2985 BYE
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0


RECEIVE << 193.220.101.242:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKFCF7BB8465974CFAA116F2BE6F28F50C
From: tommy <sip:tommy@home>;tag=2359707354
To: <sip:095128304@home>;tag=as45c544eb
Call-ID: 97D618D9-A37E-482A-B30E-5F296F5F7189@192.168.10.11
CSeq: 2985 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Content-Length: 0

x-lite log, test2 (crashing):

SEND >> 193.220.101.242:5060
INVITE sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKB41553AE0A754514967951FC27BCAA18
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7080 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 200

v=0
o=tommy 1174660792 1174660792 IN IP4 192.168.10.11
s=X-Lite
c=IN IP4 192.168.10.11
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE << 193.220.101.242:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKB41553AE0A754514967951FC27BCAA18
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as69480087
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7080 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Proxy-Authenticate: Digest realm="asterisk", nonce="776a77c8"
Content-Length: 0


SEND >> 193.220.101.242:5060
ACK sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKB41553AE0A754514967951FC27BCAA18
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as69480087
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7080 ACK
Max-Forwards: 70
Content-Length: 0


SEND >> 193.220.101.242:5060
INVITE sip:095128304@home SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK27BCC66FD948477DA092BE263D3177F6
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7081 INVITE
Proxy-Authorization: Digest username="tommy",realm="asterisk",nonce="776a77c8",response="a10b3424e176c703027588efd2e8a49e",uri="sip:095128304@home"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 200

v=0
o=tommy 1174660822 1174660822 IN IP4 192.168.10.11
s=X-Lite
c=IN IP4 192.168.10.11
t=0 0
m=audio 8000 RTP/AVP 8 101
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE << 193.220.101.242:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK27BCC66FD948477DA092BE263D3177F6
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7081 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Content-Length: 0


RECEIVE << 193.220.101.242:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK27BCC66FD948477DA092BE263D3177F6
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7081 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:095128304@193.220.101.242>
Content-Type: application/sdp
Content-Length: 195

v=0
o=root 13021 13021 IN IP4 193.220.101.242
s=session
c=IN IP4 193.220.101.242
t=0 0
m=audio 15186 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

SEND >> 193.220.101.242:5060
ACK sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bKD363F2CBA77645E5B967D3CA3BA314BD
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7081 ACK
Max-Forwards: 70
Content-Length: 0


SEND >> 193.220.101.242:5060
BYE sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6047AC11358C4BDEB5E9DDF06F16669F
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7082 BYE
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0


SEND >> 193.220.101.242:5060
BYE sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6047AC11358C4BDEB5E9DDF06F16669F
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7082 BYE
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0


SEND >> 193.220.101.242:5060
BYE sip:095128304@193.220.101.242 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.11:5060;rport;branch=z9hG4bK6047AC11358C4BDEB5E9DDF06F16669F
From: tommy <sip:tommy@home>;tag=578954439
To: <sip:095128304@home>;tag=as45c544eb
Contact: <sip:tommy@192.168.10.11:5060>
Call-ID: D4C23D3E-C11C-4B74-8D9D-98F525386635@192.168.10.11
CSeq: 7082 BYE
Max-Forwards: 70
User-Agent: X-Lite build 1101
Content-Length: 0

Comments:By: Brian West (bkw918) 2004-01-11 17:41:43.000-0600

This isn't a bug in asterisk you need to contact the asterisk-oh323 developers as this isn't included in the asterisk distro and is considered 3rd party software.  I use chan_h323 daily in production which is included with asterisk without problems.
By: digger_one (digger_one) 2004-01-11 17:46:12.000-0600

Will do that. I will do some tests with chan_h323 as well.
By: Brian West (bkw918) 2004-01-11 17:47:33.000-0600

Also openh323 and pwlib work on redhat8,9 but it doesn't work on gentoo and some other platforms.  I do have it working on slackware.   But your mileage will vary.