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

Tue Dec 15 19:41:31 CST 2009

A NOTE has been added to this issue. 
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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:                   major
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:              2009-12-15 19:41 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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---------------------------------------------------------------------- 
 (0115313) falves11 (reporter) - 2009-12-15 19:41
 https://issues.asterisk.org/view.php?id=16443#c115313 
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I found the problem. My peer's host definition had an extra period at the
right of the IP address. Nevertheless, Asterisk should not have attempted
to loookup or validate the validity of the IP address, since it can be also
a name that will be translated later, unless this is done by design.

Please advice. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-12-15 19:41 falves11       Note Added: 0115313                          
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