PJSIP Configuration For AWS/EC2 Based Asterisk 13.1.0
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 [endpoint_internal](!) [auth_userpass](!) [aor_dynamic](!) ;; 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.
local_net2.31.32.0/20
; In the following two lines, replace “
; curl -s http://169.254.169.254/latest/meta-data/public-ipv4
external_media_address=
type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no
type=auth auth_type=userpass
type=aor max_contacts=1
remove_existing=yes
;Definitions for our phones, using the templates above
2 thoughts on - PJSIP Configuration For AWS/EC2 Based Asterisk 13.1.0
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_net2.31.32.0/20
;; For softphones within EC2
local_net2.168.1.0/24
external_media_address=
external_signaling_address=
;Templates for the necessary config sections
[endpoint_internal](!)
type=endpoint
context=from-internal
disallow=all
allow=!all,ulaw
direct_media=no
rtp_symmetric=yes
BTW, the allow=!all is equivalent to disallow=all, so you can drop the disallow line.