[asterisk-bugs] [JIRA] (ASTERISK-24400) ooh323 sends wrong hangup code
Dmitry Melekhov (JIRA)
noreply at issues.asterisk.org
Tue Oct 14 22:43:29 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24400?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223029#comment-223029 ]
Dmitry Melekhov commented on ASTERISK-24400:
--------------------------------------------
Hello!
With this patch I get right busy ( i.e. code 17 ) tone , with debug and with no debug, with progress and no progress- always right result.
About channel reliability- this is leased channel, not very long- in the same town, 100 Mbit , theoretically, just because we share the same switches network of the same provider, and it is encrypted by linux ipsec on both sides, but our monitoring doesn't show us any loss , there are no complains from users, so we consider it as reliable enough.
Btw, I just reproduced the same problem with busy tone on another asterisk, which is connected to avaya definity, just nobody complained :-) It is connected by different channel, only same part is the same cisco gateway...
Thank you very much!
> ooh323 sends wrong hangup code
> ------------------------------
>
> Key: ASTERISK-24400
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24400
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Addons/chan_ooh323
> Affects Versions: 11.10.2
> Environment: centos x86-64
> Reporter: Dmitry Melekhov
> Assignee: Alexander Anikin
> Severity: Minor
> Attachments: 24400-test-2.patch, 24400-test-3.patch, 24400-test-4.patch, 24400-test.patch, cisco_log, cisco_log_tcp, debug.dump, debug-noprogress.dump, h323_log, h323_log, h323_log, h323_log, h323_log, nodebug.dump, nodebug-noprogress.dump, ooh323.conf
>
>
> Hello!
> We installed yet another asterisk which works as voip gateway between panasonic kx-ta 100 and our h323 network.
> Connection between asterisk and panasonic is ISDN PRI, namely qsig.
> For some reason we have problems with generating ring back tone from panasonic , so I added Progress, so it looks like:
> {noformat}
> exten => 5880,1,SET(FAXOPT(t38gateway)=yes)
> exten => 5880,n,Progress
> exten => 5880,n,Dial(DAHDI/g1/${EXTEN})
> exten => 5880,n,Hangup
> {noformat}
> but, in this case we get wrong hangup tone, like not user busy, but network congestion.
> If there is no Progress, then no ringback, but tone is right.
> Really, whole connection scheme is:
> {noformat}
> kt-ta100---asterisk--cisco3845--ts004-avaya
> {noformat}
> I'm avaya user :-)
> So, what I see is on asterisk:
> {noformat}
> -- Called DAHDI/g1/5880
> -- Span 1: Channel 0/2 got hangup, cause 17
> {noformat}
> on cisco:
> {noformat}
> Oct 8 03:27:58.138: //1355041/EE3142978C7C/CCAPI/ccCallDisconnect:
> Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)
> {noformat}
> I placed call through another asterisk, which is connected to ts004, over ISDN, call is over ISDN (back to ts004, etc..) , and see there:
> {noformat}
> -- Span 1: Channel 0/3 got hangup request, cause 38
> -- DAHDI/i1/5880-21aa is circuit-busy
> {noformat}
> I don't completely sure how to collect all data, because if I turn on debug or trace then I got different behaviour, namely I don't hear ringback tone even if Progress is in place.
> So I need help from developer to collect debug info.
> I guess that problem is in ooh323 because if I use sip to call asterisk from cisco3845 , then I get user busy tone.
> Thank you!
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