[asterisk-bugs] [JIRA] (ASTERISK-21194) chan_sip can fail to find a peer during reload

[Jaco Kroon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21194?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=213384#comment-213384 ] 

Jaco Kroon commented on ASTERISK-21194:
---------------------------------------

Re-looked at the code, and it's above my skill level (well, I actually just don't know the asterisk internals well enough, but from the looks of it we have to remove the object from the  ao2 container and then re-add it, which is what causes the problem).  In the meantime I've managed to reduce my reloads quite a bit and then only reload portions of asterisk as per required (ok fine, I still do a full asterisk reload but managed to prevent the config file timestamps from updating unless the content of the config files actually change, so most of the time the individual reloads just short-out due to the config file not changing) so my risk of this has also reduced significantly.  Also managed to track the issue with stopping to service the udp port to a combination of DNS time-outs (which is outside of control, and non-trivial to fix but shouldn't affect 12) and blocking of astdb (sqlite) which I've since moved onto ramfs (I don't care if it goes lost, it's SIP and IAX/2 registry information only in there), DNS is highly sporadic and mostly cached, so honestly, for me the risk of this triggering again is incredibly low now, to the point where I'd rather address other concerns.  If someone else would like to take a wack at this, please feel free, or close, however you see fit.
                
> chan_sip can fail to find a peer during reload
> ----------------------------------------------
>
>                 Key: ASTERISK-21194
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21194
>             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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