[asterisk-bugs] [Asterisk 0016443]: chan sip removes peers like if srvlookup were active, but it is not

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=16443 
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Reported By:                falves11
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16443
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   trivial
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.28-rc1 
JIRA:                       SWP-550 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 232581 
Request Review:              
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Date Submitted:             2009-12-15 00:10 CST
Last Modified:              2009-12-17 09:10 CST
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Summary:                    chan sip removes peers like if srvlookup were
active, but it is not
Description: 
These are my "sip show settings"
Global Settings:
----------------
  SIP Port:               5060
  Bindaddress:            0.0.0.0
  Videosupport:           No                                              
                                                                           
                                             
  AutoCreatePeer:         Yes
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Promsic. redir:         No
  SIP domain support:     No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Our auth realm          minixel.com
  Realm. auth:            No
  Always auth rejects:    No
  Call limit peers only:  No
  Direct RTP setup:       No
  User Agent:             Asterisk
  MWI checking interval:  10 secs
  Reg. context:           (not set)
  Caller ID:              0000000000
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  T38 fax pt UDPTL:       Yes
  RFC2833 Compensation:   Yes
  SIP realtime:           Disabled

Global Signalling Settings:
---------------------------
  Codecs:                 0x105 (g723|ulaw|g729)
  Codec Order:            g729:20,ulaw:20,g723:30
  T1 minimum:             1500
  No premature media:     No
  Relax DTMF:             Yes
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            30 
  RTP Hold Timeout:       60 
  MWI NOTIFY mime type:   text/plain
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs                                       
                                                                           
                                             
  Reg. default duration:  3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No 

Default Settings:
-----------------
  Context:                inbound
  Nat:                    Always                                          
                                                                           
                                             
  DTMF:                   auto
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               en
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk


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Issue History 
Date Modified    Username       Field                    Change               
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2009-12-17 09:10 lmadsen        Severity                 major => trivial    
2009-12-17 09:10 lmadsen        Description Updated                          
2009-12-17 09:10 lmadsen        Additional Information Updated                  
 
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