Hello,
Am trying to implement a local voip communication setup.
My scenario is
1.asterisk version 12.04
2.Trying to establish call between WebRTC(using sipml5) and Xlite
3.No internet available (so tried a local stun setup)
Problem
When calling from xlite client call is geting disconected on webside(Message: call rejected)
When calling from web to xlite,call is established but no voice.
Users are 1060(web) & 1065 (xlite).
configs
/etc/asterisk/sip.conf
[general]
context=default
bindaddr=0.0.0.0:5060
nat=force_rport,comedia
realm=192.168.1.10
transport=udp,ws,wss
allow=all
srvlookup=no
[1060]
type=peer
username=1060
host=dynamic
secret=1060
context=default
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
dtlsenable=yes
dtlsverify=no
[1065]
context=default
type=peer
username=1065
host=dynamic
secret=1065
hassip = yes; enable ice if supported by your ua (e.g. xlite)
icesupport = yes
Please help me to solve this issue