On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.geis at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20221006/2a3573fc/attachment.html>
On Thu, Oct 6, 2022 at 10:03 AM Jerry Geis <jerry.geis at gmail.com> wrote:> > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.geis at gmail.com> 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. >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. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20221006/bb6062e6/attachment.html>
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis <jerry.geis at gmail.com> wrote:> > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.geis at gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20221006/303346bf/attachment.html>