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>
On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis <jerry.geis at gmail.com> wrote:> > > 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 ? >The sample configuration file outlines how things work, and the options for it: https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 in general localnet and externip (or externaddr, or externhost) -- 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/20221006/7323252b/attachment.html>
>The sample configuration file outlines how things work, and the options for >it: >https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 >in general localnet and externip (or externaddr, or externhost)I added: externip=xxx nat=force_rport,comedia to the general section of sip.conf its still sending to the local IP. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20221006/275499b2/attachment.html>