[asterisk-bugs] [JIRA] (ASTERISK-22891) One way audio if dialplan_exec menu option runs Dial application and certain codecs are used

[Stefan Tichy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22891?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212164#comment-212164 ] 

Stefan Tichy commented on ASTERISK-22891:
-----------------------------------------

Thanks for the explanation. I supposed that using Dial could cause problems, but until now I was not sure about it. Maybe the documentation in confbridge.conf could be adjusted. 
                
> One way audio if dialplan_exec menu option runs Dial application and certain codecs are used
> --------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22891
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22891
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_confbridge
>    Affects Versions: 11.6.0
>         Environment: Debian Wheezy, i386
>            Reporter: Stefan Tichy
>            Severity: Minor
>
> The scenario is similar to the "using dialplan_exec" example in "Asterisk - The Definitive Guide". Here Dial is used instead of Originate.
> The user menu in confbridge.conf is extended by two lines. Everything else is default. 
> {code}
> *5=dialplan_exec(cbexec,test,1)
> 5=dialplan_exec(cbexec,test,1)
> {code}
> The referenced dialplan snippet is:
> {code}
> context cbexec {
>     test => {
>         Dial(SIP/25,15,gHF(conference,100,1));
>     }
> }
> {code}
> First participant (sip channel) enters the conference and dials "5" to contact another user and to fetch this user into the conference. If for example both phones use codec alaw, the following warning is printed:
> {code}
> channel.c:5081 ast_write: Codec mismatch on channel SIP/snom360-00000028 setting write format to slin from alaw native formats (alaw)
> {code}
> After this the first participant can listen to the conference but cannot speak to the conference.
> If the first phone uses gsm codec instead of alaw, everything works as expected.
> ASTERISK-21144 describes another one way audio problem. The jitterbuffer setting does not matter for the problem described here.

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