I have a problem with asterisk srtp and tls when i calling i only can hear the noise
i need help to fix in order to calling normally with asterisk srtp & tls
this is the detail
server = asterisk 1.8.23 with srtp and tls
client = eyeBeam 1.5.19
bria 2.4
Phoner Lite 2.11
sip.conf
;Asterisk Configuration
[general]
context=voip-phones ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
disallow=all ; First disallow all codecs
allow=alaw ; Allow codecs in order of preference
allow=ulaw ; Allow codecs in order of preference
transport=udp ; Set the default transports. The order determines the primary default transport.
tcpenable=no ; Enable server for incoming TCP connections (default is no)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.
bindaddr=154.20.7.10:5061 ; Listen on a specific IPv4 address.
;
; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
; “udpbindaddr”, “tcpbindaddr”, and “tlsbindaddr”.)
; Using bindaddr will only enable UDP support in order to be backwards compatible with those systems
; that were upgraded prior to TCP support. Use udpbindaddr and tcpbindaddr to bind to UDP and TCP
; independently.
encryption=yes
transport=tls
tlsenable=yes
tlsbindaddr=154.20.7.10:5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
;
[1001]
type=friend
context=voip-phones
secret=2001
host=dynamic
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
encryption=yes
transport=tls
[1002]
type=friend
context=voip-phones
secret=2002
host=dynamic
disallow=all
allow=alaw
allow=ulaw
allow=all
dtmfmode=rfc2833
encryption=yes
trasnport=tls
extensons.conf
[voip-phones]
exten => 1001,1,Dial(SIP/1001/10)
exten => 1001,2,Hangup
exten => 1002,1,Dial(SIP/1002/10)
exten => 1002,2 Hangup
anything wrong with my configuration