[asterisk-bugs] [JIRA] (ASTERISK-21194) chan_sip can fail to find a peer during reload
Jaco Kroon (JIRA)
noreply at issues.asterisk.org
Wed Apr 3 07:49:01 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-21194?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204906#comment-204906 ]
Jaco Kroon commented on ASTERISK-21194:
---------------------------------------
Been sitting on this ... fixing other more critical * issues atm (crashes are worse than call failures). Haven't forgotten.
> 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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