[asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04
Joshua C. Colp
jcolp at sangoma.com
Fri Feb 4 11:54:00 CST 2022
On Fri, Feb 4, 2022 at 1:42 PM Jerry Geis <jerry.geis at gmail.com> wrote:
>
> So - still on this...
>
> I was just dialing the SIP Gateway with Dial(SIP/103)
>
> if I change my Dial command to this:
>
> Dial(SIP/103,20,D(15))
> So I send out the DTMF in the dial command - this works and connects me
> and the DTMF is delivered and I get the right port.
>
> The problem still remains - Dialing just Dial(SIP/103) from the polycom
> phone - and then doing 15 for DTMF does not work. Cant figure out why ?
>
> Any thoughts ?
>
The usage of D(15) causes Asterisk to produce RTP on its own. Without it,
it merely forwards RTP. If a NAT/firewall requires media to be sent before
allowing media in, then you'll have no media flow. You can use the
"rtpkeepalive" option to have the RTP stack produce keepalive packets,
which will then open the NAT/firewall.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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