search for: aor_dynamic

...d now, I think. My pjsip.conf is now [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=192.168.1.0/24 ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw [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 [demo-alice](endpoint_internal) auth=demo-alice aors=demo-alice mailboxes=box_a rewrite_contact=yes [demo-alice](auth_userpass) password=demo-alice ; put a strong, unique password here instead us...

...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...

...tocol=udp > bind=0.0.0.0 > local_net=192.168.1.0/24 > ;Templates for the necessary config sections > > [endpoint_internal](!) > type=endpoint > context=from-internal > disallow=all > allow=ulaw > > [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 > > [demo-alice](endpoint_internal) > auth=demo-alice > aors=demo-alice > mailboxes=box_a > rewrite_contact=yes > [demo-alice](auth_userpass) > password...

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

...ternal_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...

...k. Here's my PJSIP.conf: [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 ... [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes message_context=text-context [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;;; Configuration for user 99999999 ... I used basically exactly the same procedure for UDP based registration and it works just fine. Any help is appreciated. The specific error I see on the CLI is: Connected to Asterisk 13.6.0 currently running...