[asterisk-bugs] [JIRA] (ASTERISK-21214) chan_sip can fail to find a peer during reload

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-21214:
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             Summary: chan_sip can fail to find a peer during reload
                 Key: ASTERISK-21214
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21214
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/General
    Affects Versions: 11.2.1
            Reporter: Jaco Kroon
            Assignee: Jaco Kroon


During a global system reload I saw this:

{noformat}
[Feb 28 16:50:26] VERBOSE[2712][C-0000317a] pbx.c:     -- Executing [number at prov:5] Dial("Local/number at foo-0000377b;2", "SIP/bar/number,,") in new stack
[Feb 28 16:50:26] VERBOSE[2712][C-0000317a] netsock2.c:   == Using SIP RTP CoS mark 5
[Feb 28 16:50:26] ERROR[2712][C-0000317a] netsock2.c: getaddrinfo("bar", "(null)", ...): Name or service not known
[Feb 28 16:50:26] WARNING[2712][C-0000317a] chan_sip.c: No such host: bar
[Feb 28 16:50:26] WARNING[2712][C-0000317a] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
{noformat}

sip show peer (after reload):

{noformat}
  * Name       : bar
  Description  : 
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : uls-makecall
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     : 
  Tonezone     : <Not set>
  Accountcode  : bar
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Named Callgr : 
  Nam. Pickupgr: 
  MOH Suggest  : 
  Mailbox      : 
  VM Extension : 8579
  LastMsgsSent : 0/0
  Call limit   : 2147483647
  Max forwards : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : no
  Force rport  : Auto (No)
  Symmetric RTP: No
  ACL          : No
  DirectMedACL : No
  T.38 support : Yes
  T.38 EC mode : Redundancy
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 10.0.0.14
  Addr->IP     : 10.0.0.14:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Reg. exten   : 
  Def. Username: 
  SIP Options  : (none)
  Codecs       : (g729)
  Codec Order  : (g729:20)
  Auto-Framing :  No 
  Status       : OK (1 ms)
  Useragent    : 
  Reg. Contact : 
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Variables    :
                 __noivr = yes
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   : 
  Use Reason   : No
  Encryption   : No
{noformat}

And it would have looked exactly the same just before reload.  The section in sip.conf:

{noformat}
[bar]
type=friend
host=10.0.0.14
qualify=yes
disallow=all
allow=g729
context=uls-makecall
directmedia=no
dtmfmode=rfc2833
accountcode=IS
jbforce=no
setvar=__noivr=yes
transport=udp
{noformat}

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