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Summary:ASTERISK-14550: SIP crash on ACK
Reporter:Jean-Louis Noel (jln17)Labels:
Date Opened:2009-07-28 09:56:47Date Closed:2011-06-07 14:00:38
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Registration
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:At the end of the registration process crash just when the connection is established on sending ACK.

==========
[Jul 27 17:18:10] DEBUG[15417]: channel.c:2986 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan->timingfd=34)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:9651 add_sdp: Done building SDP. Settling with this capability: 0x100 (g729)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:3016 initialize_initreq: Initializing
already initialized SIP dialog 47599549603048a3397eaaf11e4565a1@212.68.197.108(pr esumably reinvite)
[Jul 27 17:18:10] DEBUG[15417]: chan_sip.c:3289 __sip_xmit: Trying to put 'INVITE sip' onto UDP socket destined for 77.72.169.129:5060
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3754 __sip_ack: Acked pending invite104
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3791 __sip_ack: Stopping retransmission on '47599549603048a3397eaaf11e4565a1@212.68.197.108' of Request 104: Match Found
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:16796 handle_response_invite: SIP response 200 to RE-invite on outgoing call 47599549603048a3397eaaf11e4565a1@212.68.197.108
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:7625 process_sdp: SDP version number same as previous SDP. Not parsing this SDP.
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:5282 update_call_counter: Updating call counter for outgoing call
[Jul 27 17:18:10] DEBUG[14068]: chan_sip.c:3289 __sip_xmit: Trying to put 'ACK sip:+3' onto UDP socket destined for 77.72.169.129:5060
ns*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[root@ns asterisk]#
==========
Nothing happend if I set canreinvite to no.

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

<------------>
   -- Native bridging SIP/2100-08c67448 and SIP/VoipCheap-08c72b58
set_destination: Parsing <sip:+3271382209@77.72.169.129:5060> for address/port t
o send to
set_destination: set destination to 77.72.169.129, port 5060
Audio is at 212.68.197.108 port 19612
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.72.169.129:5060:
INVITE sip:+3271382209@77.72.169.129:5060 SIP/2.0
Via: SIP/2.0/UDP 212.68.197.108:5060;branch=z9hG4bK30f257c6;rport
Max-Forwards: 70
From: "Jean-Louis" <sip:2100@212.68.197.108>;tag=as3d75b8af
To: <sip:+3271382209@77.72.169.129>;tag=110113ac4a27939d462ce9
Contact: <sip:2100@212.68.197.108>
Call-ID: 45201e3d42fc3bf25ab0c9440d5ab3e5@212.68.197.108
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.2.0-beta3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1094596987 1094596989 IN IP4 10.149.85.17
s=Asterisk PBX 1.6.2.0-beta3
c=IN IP4 10.149.85.17
t=0 0
m=audio 50000 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
ns*CLI>
<--- SIP read from UDP:77.72.169.129:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 212.68.197.108:5060;branch=z9hG4bK30f257c6;rport
From: "Jean-Louis" <sip:2100@212.68.197.108>;tag=as3d75b8af
To: <sip:+3271382209@77.72.169.129>;tag=110113ac4a27939d462ce9
Contact: sip:+3271382209@77.72.169.129:5060
Call-ID: 45201e3d42fc3bf25ab0c9440d5ab3e5@212.68.197.108
CSeq: 104 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 219

v=0
o=jln17 1248791419 1248791419 IN IP4 194.120.0.34
s=SIP Call
c=IN IP4 194.120.0.34
t=0 0
m=audio 24736 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
--- (11 headers 10 lines) ---
set_destination: Parsing <sip:+3271382209@77.72.169.129:5060> for address/port t
o send to
set_destination: set destination to 77.72.169.129, port 5060
Transmitting (no NAT) to 77.72.169.129:5060:
ACK sip:+3271382209@77.72.169.129:5060 SIP/2.0
Via: SIP/2.0/UDP 212.68.197.108:5060;branch=z9hG4bK75750e49;rport
Max-Forwards: 70
From: "Jean-Louis" <sip:2100@212.68.197.108>;tag=as3d75b8af
To: <sip:+3271382209@77.72.169.129>;tag=110113ac4a27939d462ce9
Contact: <sip:2100@212.68.197.108>
Call-ID: 45201e3d42fc3bf25ab0c9440d5ab3e5@212.68.197.108
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.2.0-beta3
Content-Length: 0


---
ns*CLI>
<--- SIP read from UDP:91.178.51.53:5061 --->
SIP/2.0 200 OK
From: "asterisk"<sip:asterisk@212.68.197.108>;tag=as00781658
To: <sip:2002@192.168.1.12:5060>;tag=c01a8c0-13c4-17725-5b95f22-4338
Call-ID: 3f3a49603ddfb58f4a068c3273355a8d@212.68.197.108
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 212.68.197.108:5060;rport=5060;branch=z9hG4bK3ddc459f
Supported: replaces,100rel,timer
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK, UPDATE
User-Agent: Swissvoice IP10 SP v1.0.1 (Build 4) 3.0.5.1
Accept: application/sdp
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3f3a49603ddfb58f4a068c3273355a8d@212.68.197.108' M
ethod: OPTIONS
ns*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Comments:By: Leif Madsen (lmadsen) 2009-08-31 08:26:11

Can you please re-test with the latest 1.6.2 branches from subversion?

(svn co http://svn.asterisk.org/svn/asterisk/branches/1.6.2 asterisk-1.6.2-vanilla)

Or you can wait for Asterisk-1.6.2.0-rc1 which should be released early this week.

If you get the same crash issue, please attach a backtrace per the backtrace.txt file in the doc/ subdirectory of your Asterisk source. Additionally, make sure you have enabled DONT_OPTIMIZE in the Compiler Flags of menuselect.

Because this is SIP related, you need to provide the following information:

* history of the call (recordhistory=yes, dumphistory=yes, in sip.conf)
* sip debug (which you've provided)
* relative configuration and dialplan snippet in order to reproduce the issue
* backtrace of the crash

Thanks!
By: Jean-Louis Noel (jln17) 2009-09-07 04:45:39

This issue is cleared with asterisk-1.6.2-vanilla
and asterisk-1.6.2-rc1.
Thank you.
By: Olle Johansson (oej) 2009-09-07 08:13:56

THanks for reporting back to us!