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

[Matthew Jordan]

----- 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