[asterisk-bugs] [JIRA] (ASTERISK-21231) When outboundpoxy is set, asterisk should not attempt to resolve DNS

Mon Mar 11 03:05:01 CDT 2013

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21231?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=204083#comment-204083 ] 

Walter Doekes commented on ASTERISK-21231:
------------------------------------------

{{dialog->sa}} is now unset... but you're still setting stuff on it. And finally we get here when deciding what our sent-by address is for this outbound message.
{noformat}
        if (ast_connect(s, them)) {
                ast_log(LOG_WARNING, "Cannot connect\n");
                close(s);
                return -1;
        }
{noformat}

Here you stumble upon a related bug, which you might be able to solve at once:

  if you put {{belanger.home  127.0.0.1}} in your hosts file,
  you'll get 127.0.0.1 in the Via sent-by even though your
  proxy is on the "outside"

If you set {{dialog->sa}} to the outboundproxy ip:port in {{createaddr}} you might fix that second bug at the same time.

Unfortunately deciding whether you're not breaking something else at the same time is a bit trickier ;)

> When outboundpoxy is set, asterisk should not attempt to resolve DNS
> --------------------------------------------------------------------
>
>                 Key: ASTERISK-21231
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21231
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.20.1, 10.12.1, 11.2.1
>            Reporter: Paul Belanger
>         Attachments: outboundproxy_no_dns.diff, outboundproxy_no_dns.v2.diff
>
>
> Using the following sip.conf:
> {code}
> [general](+)
> nat = yes
> outboundproxy = 209.87.247.153
> srvlookup = no
> {code}
> And the following extensions.conf:
> {code}
> [ivr-polybeacon.com]
> exten => 100,1,NoOp()
>     same => n,Dial(SIP/100 at belanger.home)
> {code}
> Creates the following full log:
> {code}
> [2013-03-08 19:25:44.715] DEBUG[23027] pbx.c: Launching 'NoOp'
> [2013-03-08 19:25:44.715] VERBOSE[23027] pbx.c:     -- Executing [100 at ivr-polybeacon.com:1] NoOp("SIP/kamailio-01-prod-00000020", "") in new stack
> [2013-03-08 19:25:44.715] DEBUG[23027] pbx.c: Launching 'Dial'
> [2013-03-08 19:25:44.715] VERBOSE[23027] pbx.c:     -- Executing [100 at ivr-polybeacon.com:2] Dial("SIP/kamailio-01-prod-00000020", "SIP/100 at belanger.home") in new stack
> [2013-03-08 19:25:44.715] CC[23027] ccss.c: Agent policy for SIP/kamailio-01-prod-00000020 is 'never'. CC not possible
> [2013-03-08 19:25:44.715] DEBUG[23027] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
> [2013-03-08 19:25:44.715] DEBUG[23027] chan_sip.c: Allocating new SIP dialog for 177e5ce53b54f4320ad6f1113b311316 at 209.87.247.153:5062 - INVITE (No RTP)
> [2013-03-08 19:25:44.715] DEBUG[23027] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x1af47e8'
> [2013-03-08 19:25:44.715] DEBUG[23027] res_rtp_asterisk.c: Allocated port 13420 for RTP instance '0x1af47e8'
> [2013-03-08 19:25:44.715] DEBUG[23027] rtp_engine.c: RTP instance '0x1af47e8' is setup and ready to go
> [2013-03-08 19:25:44.715] DEBUG[23027] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x1af47e8'
> [2013-03-08 19:25:44.715] VERBOSE[23027] netsock2.c:   == Using SIP RTP CoS mark 5
> [2013-03-08 19:25:44.715] DEBUG[23027] chan_sip.c: Setting NAT on RTP to On
> [2013-03-08 19:25:44.715] DEBUG[23027] chan_sip.c: OBPROXY: Applying global OBproxy to this call
> [2013-03-08 19:25:44.716] DEBUG[23027] netsock2.c: Splitting 'belanger.home' into...
> [2013-03-08 19:25:44.716] DEBUG[23027] netsock2.c: ...host 'belanger.home' and port ''.
> [2013-03-08 19:25:44.717] ERROR[23027] netsock2.c: getaddrinfo("belanger.home", "(null)", ...): Name or service not known
> [2013-03-08 19:25:44.717] WARNING[23027] chan_sip.c: No such host: belanger.home
> [2013-03-08 19:25:44.717] DEBUG[23027] chan_sip.c: Cant create SIP call - target device not registered
> [2013-03-08 19:25:44.717] DEBUG[23027] chan_sip.c: Destroying SIP dialog 177e5ce53b54f4320ad6f1113b311316 at 209.87.247.153:5062
> [2013-03-08 19:25:44.717] VERBOSE[23027] chan_sip.c: Really destroying SIP dialog '177e5ce53b54f4320ad6f1113b311316 at 209.87.247.153:5062' Method: INVITE
> [2013-03-08 19:25:44.717] DEBUG[23027] rtp_engine.c: Destroyed RTP instance '0x1af47e8'
> [2013-03-08 19:25:44.718] WARNING[23027] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
> [2013-03-08 19:25:44.718] VERBOSE[23027] app_dial.c:   == Everyone is busy/congested at this time (1:0/0/1)
> {code}

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