[asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833
nachum.yaron at gmail.com
Tue Mar 11 12:16:14 CDT 2014
Thanks Mathew. It's good to know the limitations :-)
Is there any plan to add it?
On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan <mjordan at digium.com> wrote:
> On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum <nachum.yaron at gmail.com>
> > Hi Mathew,
> > The regular sip stack has 'auto' dtmfmode which behaved as I said - if
> > remote replied with telephony event it used RFC2833 otherwise it used
> > inband.
> Correct. There is no setting for dtmf_mode that is analogous to the
> chan_sip 'auto' setting - what you configure for you endpoint today is
> what it will use.
> That's not a bug, just something not existing yet.
> 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
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-users