[asterisk-bugs] [Asterisk 0012494]: asterisk locks after p2p sip channel bridge

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12494 
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Reported By:                pj
Assigned To:                murf
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Project:                    Asterisk
Issue ID:                   12494
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 114536 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             04-22-2008 13:22 CDT
Last Modified:              04-23-2008 14:10 CDT
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Summary:                    asterisk locks after p2p sip channel bridge
Description: 
simple call between two sip phones (both have same codec), 
console log and 'core show locks' attached
this bug was probably caused after huge commits in rev 114190,
it happens in 100% of sip calls, when p2p bridge is attempted, 
so it's really big issue!
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---------------------------------------------------------------------- 
 pj - 04-23-08 14:10  
---------------------------------------------------------------------- 
another weird behaviour:
even when peer is successfully registered, calls placed from this phone
are not placed from peer context, but are placed from context specified in
[general] section in sip.conf (here I have from-guest context)

ipbx*CLI> sip show peer 324p

  Addr->IP     : 193.85.164.154 Port 11676
  Defaddr->IP  : 0.0.0.0 Port 5060
  Transport    : UDP
  Def. Username:
  SIP Options  : (none)
  Codecs       : 0x502 (gsm|g729|ilbc)
  Codec Order  : (g729:20,gsm:20,ilbc:30)
  Auto-Framing :  No
  100 on REG   : No
  Status       : OK (371 ms)
  Useragent    : RTC/1.5.5374
  Reg. Contact : sip:10.0.0.3:12654



[Apr 23 21:13:26]   == Using SIP RTP CoS mark 5
[Apr 23 21:13:26] Sending to 193.85.164.154 : 11676 (NAT)
[Apr 23 21:13:26] Using INVITE request as basis request -
0fa50eb50000000000f0172176a5c801
[Apr 23 21:13:26] No user '324p' in SIP users list
[Apr 23 21:13:26] No matching peer for '324p' from '193.85.164.154:11676'
[Apr 23 21:13:26] Found RTP audio format 8
[Apr 23 21:13:26] Found RTP audio format 0
[Apr 23 21:13:26] Found RTP audio format 3
[Apr 23 21:13:26] Found RTP audio format 97
[Apr 23 21:13:26] Found RTP audio format 101
[Apr 23 21:13:26] Peer audio RTP is at port 193.85.164.154:11698
[Apr 23 21:13:26] Found audio description format GSM for ID 3
[Apr 23 21:13:26] Found audio description format PCMA for ID 8
[Apr 23 21:13:26] Found audio description format PCMU for ID 0
[Apr 23 21:13:26] Found unknown media description format RED for ID 97
[Apr 23 21:13:26] Found audio description format telephone-event for ID
101
[Apr 23 21:13:26] Capabilities: us - 0x50a (gsm|alaw|g729|ilbc), peer -
audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xa (gsm|alaw)
[Apr 23 21:13:26] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Apr 23 21:13:26] Peer audio RTP is at port 193.85.164.154:11698
[Apr 23 21:13:26] Looking for 959 in from-guest (domain ipbx.i.cz)
[Apr 23 21:13:26] list_route: hop: <sip:10.0.0.3:12654>
[Apr 23 21:13:26]
<--- Transmitting (NAT) to 193.85.164.154:11676 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.0.0.3:12654;branch=z9hG4bKadd9b7fc89-0;received=193.85.164.154
From: "HTC_TyTN" <sip:324p at ipbx.i.cz>;tag=94e3dfc76b;epid=9b9e52b330
To: <sip:959 at ipbx.i.cz>
Call-ID: 0fa50eb50000000000f0172176a5c801
CSeq: 1 INVITE
Server: Asterisk PBX SVN-trunk-r114595
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:959 at 193.179.38.20>
Content-Length: 0


<------------>
[Apr 23 21:13:26]     -- Executing [959 at from-guest:1]
Set("SIP/ipbx.i.cz-082a5e58", "CALLERID(name)=HTC_TyTN (nreg)") in new
stack 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-23-08 14:10  pj             Note Added: 0085907                          
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