[asterisk-users] SRTP woes

[John Cahill]

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Hi,

I'm running Asterisk 11.3.0 on wheezy.
I'm trying to do TLS +SRTP with blink SIP clients as shown here
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial.

TLS is fine and I can call between clients. SRTP is a different matter,
my SIP clients return: SIP 488 "Not acceptable Here"

I'm really stumped on this one, any ideas?

srtp module is loaded:
*CLI> module show like res_srtp.so
Module                         Description                             
Use Count
res_srtp.so                    Secure RTP (SRTP)                       
0        
1 modules loaded


extensions.conf extract

exten => 1002,1,Set(_SIPSRTP=${SIPPEER(1002,srtpcapable)})
exten => 1002,n,Set(CHANNEL(secure_bridge_signaling)=1)
exten => 1002,n,Dial(SIP/exten1002,20)
exten => 1002,n,Hangup()

sip.conf extract:

[exten1002]
type=friend
host=dynamic
secret=averygoodone
context=users
nat=force_rport,comedia
encryption=yes
transport=tls

Thanks,

Regards,
John
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