Re: Trying asterisk on AWS
Jerry Geis <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CABr8-B7S05yiPtpy1gYFta-W3DQkFXR=sS7YSC3Yk47tnU0u7A@mail.gmail.com> |
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis <[email protected]> wrote: > > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <[email protected]> wrote: > >> I am trying to get audio to work on AWS using asterisk 18.14.0 >> >> I have enabled the firewall to allow ALL UDP on AWS >> >> My SIP extension has >> nat=force_rport,comedia >> qualify=yes >> allow=ulaw >> allow=alaw >> allow=gsm >> canreinvite=yes >> >> I enable "rtp set debug on" and the console is printing info. >> >> The call comes into my linphone softphone - but I get no audio on my >> linphone softphone. >> What might I be missing to allow the audio ? >> Volume is up. >> >> Thanks >> >> Jerry >> > > > I just noticed the RTP log is sending to 192.168.2.0 which is my local lan > address of the linphone - it should be sending to the NAT address and is > not. > What did I not set correctly ? > I am not using pjsip - but the older asterisk. > > Thanks > > Jerry > >Have you configured chan_sip to know it is behind NAT itself and what its >public IP address is? If not, then you'll get no audio. I'm thinking I have not. What did I miss ? Thanks, Jerry -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users