BTW, the allow=!all is equivalent to disallow=all, so you can drop the disallow line.
On Thu, Mar 5, 2015 at 7:26 PM, Sonny Rajagopalan < [email protected]> wrote: > OK. I think I found the issue. > > The key is to add > > rtp_symmetric=yes > > Here's what my final configuration looks like: > > [transport-udp] > > type=transport > > protocol=udp > > bind=0.0.0.0 > > ;; for within EC2 > > local_net=172.31.32.0/20 > > ;; For softphones within EC2 > > local_net=192.168.1.0/24 > > external_media_address=<publicIPOfEC2Instance> > > external_signaling_address=<publicIPOfEC2Instance> > > ;Templates for the necessary config sections > > > [endpoint_internal](!) > > type=endpoint > > context=from-internal > > disallow=all > > allow=!all,ulaw > > direct_media=no > > rtp_symmetric=yes > > > > On Thu, Mar 5, 2015 at 5:52 PM, Sonny Rajagopalan < > [email protected]> wrote: > >> Hello All, >> >> I have an Asterisk server v13.1.0 running on EC2 and I am able to connect >> and register SIP devices and "see" them on the asterisk CLI. I am also able >> to place calls, but I am not able to hear any audio on either end after the >> call is picked up. >> >> I was wondering if you can tell me what a minimal configuration for >> Asterisk on EC2 looks like. My current pjsip.conf configuration looks >> like this: >> >> type=transport >> protocol=udp >> bind=0.0.0.0 >> local_net=172.31.32.0/20 >> ; In the following two lines, replace "<publicIP>" with the output of >> ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 >> external_media_address=<publicIP> >> external_signaling_address=<publicIP> >> >> [endpoint_internal](!) >> type=endpoint >> context=from-internal >> disallow=all >> allow=ulaw >> direct_media=no >> >> [auth_userpass](!) >> type=auth >> auth_type=userpass >> >> [aor_dynamic](!) >> type=aor >> max_contacts=1 >> remove_existing=yes >> ;Definitions for our phones, using the templates above >> >> ;; usernames and passwords etc. below >> >> >> My security group configuration allows TCP, UDP posrt 5060 inbound, >> outbound from/to 0.0.0.0/0 and TCP, UDP ports 10000-20000 from/to >> 0.0.0.0/0. >> >> Should I turn on STUN for my zoiper softphones? Any specific flavor? >> >> What am I doing wrong? Any help appreciated. >> >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
