On Thu, Oct 6, 2022 at 10:03 AM Jerry Geis <jerry.g...@gmail.com> wrote:
> > > On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.g...@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
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