[asterisk-bugs] [JIRA] (ASTERISK-25095) Not manage "busy/congestion" in Asterisk release after v. 1.8.29.0

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25095?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226228#comment-226228 ] 

Rusty Newton commented on ASTERISK-25095:
-----------------------------------------

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> Not manage "busy/congestion" in Asterisk release after v. 1.8.29.0
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-25095
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25095
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 1.8.32.3, 13.3.2
>         Environment: Ubuntu 14.04 LTS  (64 bit)
>            Reporter: Top
>
> Hi all,
> After upgrading to the last version of Asterisk (1.8.32.3) when I call a number to CCM (from Asterisk) Asterisk don't manage the:
> "Everyone is busy/congested at this time.....Auto fallthrough, channel 'SIP/callman01-00000400' status is 'CONGESTION'" if the CCM close with a busy after some rings.
> Upgrading to the new and last branch (13.x) I've the same result.
> Downgrading to Asterisk 1.8.29.0 (my old version) it works correctly!
> Here is the verbose log:
> a) v. 1.8.29.0 
> {noformat}
> ....
> Scheduling destruction of SIP dialog '6a49af6955b2084b7b37465a5992be6e at xxx.xxx.xxx.xxx:5060' in 6400 ms (Method: INVITE)
>   == Everyone is busy/congested at this time (2:0/1/1)
> -- Auto fallthrough, channel 'SIP/callman01-00000400' status is 'CONGESTION'
> {noformat}
> ...In this case the user hear a vocal message that is related to the busy/congestion signal.
> b) v. 1.8.32.3
> {noformat}
> ...
> Scheduling destruction of SIP dialog '6e7686d8178cd38a56d8a69a14bab358 at xxx.xxx.xxx.xxx:5060' in 6400 ms (Method: INVITE)
> {noformat}
> ...In this case the Asterisk user hear ringing forever even if at the other side the CCM has returned to Asterisk the Busy/Congested signal
> ...Also the line from "sip show channels" after the "CCM close" disappear, but not the ringing/call.
> Is a new bug or it has already been solved?
> If not, could anyone patch it?
> Thanks in advance
> PS: Here a sample config:
> ----------------------------------------------------------------------
> {noformat}
> [macro-dialout-callmanager]
> exten => s,1,ChanIsAvail(SIP/callman02&SIP/callman01)
> exten => s,2,Dial(${CUT(AVAILCHAN,,1)}/${ARG1},${ARG2},${ARG3})
> exten => s,3,Hangup
> exten => s,102,Congestion
> exten => 6230,1,Answer()           <------ Asterisk user call this and this call CCM (from PHPAGI)
> exten => 6230,n,AGI(test.php)
> {noformat}
> ----------------------------------------------------------------------
> {noformat}
> [callman01]
> type=friend
> context=incoming-internal
> host=x.x.x.x
> disallow=all
> allow=alaw
> allow=ulaw
> nat=no
> canreinvite=no
> qualify=yes
> [callman02]
> type=friend
> context=incoming-internal
> host=x.x.x.x
> disallow=all
> allow=alaw
> allow=ulaw
> nat=no
> canreinvite=no
> qualify=yes
> {noformat}
> ----------------------------------------------------------------------
> test.php
> {noformat}
> #!/usr/bin/php -q
> <?php
>     error_reporting( E_ALL ^ E_NOTICE);
>     // Asterisk related
>     require('phpagi.php');
>        $agi = new AGI();
>        $agi->answer();
>        $agi->stream_file('hello');
>             $callstatus = $agi->exec_dial('SIP/callman02&SIP/callman01','3000');
>             $dialstatus = $agi->get_variable("DIALSTATUS");
>             switch ($dialstatus[data]) {
>                 case "CHANUNAVAIL" : // Channel unavailable (for example in sip.conf, when using qualify=, the SIP chan is unavailable)
>                     $agi->stream_file('busy');
>                     break;
>                 case "BUSY" : //Returned busy
>                     $agi->stream_file('busy');
>                     break;
>                 case "NOANSWER" : //No Answer (i.e SIP 480 or 604 response)
>                     $agi->stream_file('retry');
>                     break;
>                 case "ANSWER" : //Call was answered
>                     $agi->stream_file('byebye');
>                     break;
>                 case "CANCEL" : //Call attempt cancelled (i.e user hung up before the call connected)
>                     $agi->stream_file('retry');
>                     break;
>                 case "DONTCALL" : //Privacy manager don't call
>                     $agi->stream_file('retry');
>                     break;
>                 case "TORTURE" : //Privacy manager torture menu
>                     $agi->stream_file('retry');
>                     break;
>                 case "CONGESTION" : //Means Congestion, or anything else (some other error setting up the call)
>                     $agi->stream_file('retry');
>                     break;
>                 default:
>                    break;
>             }
>             break;
>     $agi->stream_file('the-end');
> ?>
> {noformat}



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