[asterisk-bugs] Asterisk not sending SIP Info with SendDTMF(F)

Rafael Morita rafael.morita at voicetechnology.com.br
Tue Sep 25 10:48:31 CDT 2012


Hello Matthew,

Actually I can send the F DTMF by RFC2833 too.
The patch in ASTERISK-17372 worked for my necessities!

Thank you very much!

On Mon, Sep 24, 2012 at 10:08 PM, Matthew Jordan <mjordan at digium.com> wrote:

>
> ----- Original Message -----
>
> > From: "Rafael Morita" <rafael.morita at voicetechnology.com.br>
> > To: asterisk-bugs at lists.digium.com
> > Sent: Monday, September 24, 2012 7:05:59 PM
> > Subject: [asterisk-bugs] Asterisk not sending SIP Info with
> > SendDTMF(F)
>
> > Hi,
>
> > I have to send a SIP Info message with flashing signal to an ATA with
> > FXO port. It will hookflash the analogic port.
> > The call is been dialed from the same ATA FXO port.
>
> > The scenario I want to get is: PBX calls FXO -> ATA send INVITE to
> > Asterisk (SIP) -> SIP Info (F DTMF) to ATA -> FXO hookflash
>
> <snip>
>
> The SIP channel driver does not support flash events.  The most that it
> will support is passing through a flash event received from another SIP
> UA.  It will not send flash events itself.
>
> <snip>
>
> > What can I do?
>
> Typically, questions such as these are best handled on the asterisk-users
> mailing list.  asterisk-bugs is typically used for commenting on a JIRA
> issue (typically to have it opened or closed).  Users on asterisk-users
> may have some suggestions for you; however, unless a patch is written that
> implements handling the AST_CONTROL_FLASH frame in chan_sip, there probably
> isn't much you can do.
>
> Patches, of course, are welcome.
>
> Note that an issue already exists for this limitation in chan_sip - see
> ASTERISK-17372 (although it does not appear as if the patches on that issue
> will fully resolve the problem either).
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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