Jerry Geis
2023-Jul-19 16:42 UTC
[asterisk-users] audio from soft phone actual phone from cloud
I have a cloud server... I have a phone in Chicago I have a phone in Indiana. Both are registered to the cloud server - using chan_sip and Asterisk 18.18.0 I can send a pre-recorded message to Chicago it auto answers and hear audio. I can do the same to the phone in indiana. however - when i call from Indiana to Chicago - the phone rings - but I do not get any audio? I have in sip.conf externip=real_ip_here localnet=172.31.17.0/255.255.255.0 localnet=192.168.11.0/255.255.255.0 localnet=192.168.1.0/255.255.252.0 localnet=10.0.0.0/255.255.255.0 One phone config: (both are the same) [YYYYY] type=friend defaultname=YYYYY defaultuser=YYYYY secret=notshown dtmfmode=RFC2833 host=dynamic description=testing. context=some-context-that-works rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid=YYYYYY qualify=yes insecurecanreinvite=yes timezone=0 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Which accounts for all locations. Why might I not be getting audio ? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230719/870206ca/attachment.html>
Frank Vanoni
2023-Aug-05 07:54 UTC
[asterisk-users] audio from soft phone actual phone from cloud
On Wed, 2023-07-19 at 12:42 -0400, Jerry Geis wrote:> Why might I not be getting audio ?Make sure the RTP port range is correctly configured and open on your server's firewall. The port range is defined in /etc/asterisk/rtp.conf The same range of UDP ports must be correctly forwarded on your firewall from the outside to Asterisk. For example, in rtp.conf: [general] ; ; RTP start and RTP end configure start and end addresses ; rtpstart=10002 rtpend=10199