Hi,
I use Asterisk 11.5.1 and it works fine. :)
Now I want to use TLS and media encryption. I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I place a call via Blink-Client (0.5.0) I get connected and Blink
shows 2 locks. The blue lock shows "Signaling is encrypted using TLS"
and the orange lock shows "Media is encrypted using sRTP". BUT i hear no
audio. After ~60 seconds I get the following message:
NOTICE[21005]: chan_sip.c:28800 check_rtp_timeout: Disconnecting call
'SIP/tgoellner-0000002c' for lack of RTP activity in 62 seconds
"sip show peers" shows me, that my Blink-Client is registered on port
60071. All other SIP-Clients (no TLS an no media encryption) are
registered at port 5060.
I tried to open the tcp and udp port range from 10000 to 61000 (in
iptables). But with no success.
I am not sure, but I think it's a firewall/NAT problem?! (Yes, my client
is behind a router > NAT)
Any idea?
-Thorsten-
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