Hello.

I try to connect Asterisk as a client to Freeswitch using TLS and SRTP.
Outbound direction (from Asterisk) - TLS/SRTP works fine.
There is an SRTP issue with inbound calls.
FS in SDP offer sends two 'm=' lines:

   m=audio 23036 RTP/SAVP 0 8 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa
   a=ptime:20
   m=audio 23036 RTP/AVP 0 8 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20

Receiving that Asterisk reply:
 WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not supported

How to solve it from Freeswitch side? I need to offer only RTP/SAVP.
Also, looking ahead, is it possible to add and modify a=crypto: line, I need to send SHA1_80 in offer or both?

Asterisk 1.8.9.2
FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)

SRTP enabled in dialplan/public.xml <action application="export" data="nolocal:sip_secure_media=true"/>

Please, advise.
Thanks!