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

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21194?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=203596#comment-203596 ] 

Michael L. Young commented on ASTERISK-21194:
---------------------------------------------

Sure... this is the documentation I looked at briefly before replying.  Not sure if it is very clear though.

{code}
/*! \brief Re-read SIP.conf config file
\note   This function reloads all config data, except for
        active peers (with registrations). They will only
        change configuration data at restart, not at reload.
        SIP debug and recordhistory state will not change
 */
static int reload_config(enum channelreloadreason reason)
{
{code}

If I am understanding the code correctly (I am sure someone will chime in if I am wrong), I see the peers are marked, the default settings are set, config settings then are set, peers are then built, if a peer is found it is retrieved and unlinked from the peers table, settings are then applied to it and then added (linked) back in to the peers table, peers that were marked and but not found during the build_peer process (user deleted it) will be unlinked at the end, then peers are poked (qualified).

So I would think that as things currently are, there is room for this to occur when the peer is not linked in the peers table while its settings are being checked and applied before being linked back in.

I am still learning chan_sip.so module's code.  We can wait for some confirmation as to whether my assessment is correct or not.
                
> 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
>
> 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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