[Freeswitch-users] How to disable two 'm=' lines

[Mihail]

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!
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