[asterisk-bugs] [JIRA] (ASTERISK-21897) D option in Dial doesn't recognize "w" as a pause

Michael L. Young (JIRA) noreply at issues.asterisk.org
Tue Jun 11 10:25:09 CDT 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-21897?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=207155#comment-207155 ] 

Michael L. Young commented on ASTERISK-21897:
---------------------------------------------

In looking at the debug log, I see that as soon as the call is answered, Asterisk sends the DTMF.  But, it appears that there is a re-invite involved.

So, I am thinking that we start to send audio to the first ip and then the re-invite occurs which changes the ip of were the audio is being sent, causing your dtmf not to work as you expect.

I am leaning towards a configuration issue.

Please look at "core show application Dial".

{noformat}
 D([called][:calling[:progress]]): Send the specified DTMF strings
    *after* the called party has answered, but before the call gets bridged.
    The <called> DTMF string is sent to the called party, and the <calling>
    DTMF string is sent to the calling party.  Both arguments can be used
    alone.  If <progress> is specified, its DTMF is sent to the called party
    immediately after receiving a PROGRESS message.
    See SendDTMF for valid digits.
{noformat}
                
> D option in Dial doesn't recognize "w" as a pause
> -------------------------------------------------
>
>                 Key: ASTERISK-21897
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21897
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_dial
>    Affects Versions: 11.4.0
>         Environment: Fedora 17
>            Reporter: Sean Darcy
>            Assignee: Sean Darcy
>         Attachments: dtmf-sip-debug-clean
>
>
> The D option in the Dial command does not recognize 'w' as a pause:
> Dial("DAHDI/1-1", "SIP/<>,,D(wwwwwww249#)") in new stack
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/<>
>     -- SIP/<>-00000063 answered DAHDI/1-1
>     -- Sending DTMF 'wwwwwww249#' to the called party.
> Interestingly, if you put other characters in the DTMF string, you get an error:
> -- Sending DTMF 'pwwwwwww249#' to the called party.
> [Jun 10 18:28:03] WARNING[20775][C-000000fc]: app.c:780 ast_dtmf_stream: Illegal DTMF character 'p' in string. (0-9*#aAbBcCdD allowed)
> This appears to be a regression since it once worked:
> https://issues.asterisk.org/jira/browse/ASTERISK-5536
> http://forums.whirlpool.net.au/archive/609344

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