On Fri, Feb 4, 2022 at 1:42 PM Jerry Geis <jerry.g...@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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