[asterisk-users] SRTP only from asterisk to extention possible

[Jakob-Matthias Böttger]

Hello,

trying to implement srtp with already working tls i somehow stuck with
srtp. If the extension has successfully registered a call from asterisk
to that extension works fine. But the other way round nothing happens.

[Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit
of 0x7fc8880467e0 (len 609) to 123.456.789:36785 returned -2: Success
[Aug 13 14:54:20] NOTICE[31053]: chan_sip.c:29623 sip_poke_noanswer:
Peer '200' is now UNREACHABLE!  Last qualify: 53

The phone (gxp1450) is configured to use symetric rtp with disabled
crypto lifetime and srtp mode forced.


Any ideas what could be wrong?

[general]
allowguest=no
alwaysauthreject=yes
nat=force_rport,comedia
sendrpid=rpid
trustrpid=yes
language=de
callevents=yes
qualify=yes
faxdetect=yes
t38pt_udptl=no
disallow=all
allow=ulaw
allow=alaw
;-------------------------Encryption-----
encryption=yes
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/var/lib/asterisk/keys/asterisk.pem
tlscafile=/var/lib/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tlsdontverifyserver=yes
;--------------------------Default----------------
context=default                 ; Default context for incoming calls
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
srvlookup=yes

[my-codecs](!)                    ; a template for my preferred codecs
disallow=all
allow=g722
allow=alaw
allow=ulaw
allow=speex
allow=g723
allow=ilbc
allow=g729
allow=gsm

[NAT](!,my-codecs)
dtmfmode=rfc2833
context=sip-out
type=friend
host=dynamic
transport=tls,tcp
qualify=yes
directmedia=no

[200](NAT)
callerid=200
defaultuser=200
fromuser=200
secret=password
mailbox=200 at default

Best Regards Jakob

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