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                <div style="height:1px;"> Thank you for replay.<br /> I tried to set up in ../conf/vars.xml<br />
                        <br /> </div>&#160;<br />
                <br />
                <br /> &lt;X-PRE-PROCESS cmd="set" data="sdp_m_per_ptime=false"/&gt;<br />
                <br /> After reloadxml and shutdown.<br /> It did not help. An issue still exist.<br />
                <br />
                <div> --- Исходное сообщение ---<br /> От кого: "Anthony Minessale" &lt;anthony.minessale@gmail.com&gt;<br /> Кому: "FreeSWITCH Users Help" &lt;freeswitch-users@lists.freeswitch.org&gt;<br /> Дата: 21 февраля 2012, 18:29:21<br /> Тема: Re: [Freeswitch-users] How to disable two 'm=' lines<br /> </div>
                <br />
                <br />
                <br />
                <blockquote class="ukr_editor_quotation" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><pre>
set the global var sdp_m_per_ptime=false in vars.xml


2012/2/21 Mihail &lt;<a href="mailto:pazuzu@ukr.net">pazuzu@ukr.net</a>&gt;:
&gt;
&gt; Hello.
&gt;
&gt; I try to connect Asterisk as a client to Freeswitch using TLS and SRTP.
&gt; Outbound direction (from Asterisk) - TLS/SRTP works fine.
&gt; There is an SRTP issue with inbound calls.
&gt; FS in SDP offer sends two 'm=' lines:
&gt;
&gt;    m=audio 23036 RTP/SAVP 0 8 101 13
&gt;    a=rtpmap:101 telephone-event/8000
&gt;    a=fmtp:101 0-16
&gt;    a=crypto:1 AES_CM_128_HMAC_SHA1_32
&gt; inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa
&gt;    a=ptime:20
&gt;    m=audio 23036 RTP/AVP 0 8 101 13
&gt;    a=rtpmap:101 telephone-event/8000
&gt;    a=fmtp:101 0-16
&gt;    a=ptime:20
&gt;
&gt; Receiving that Asterisk reply:
&gt;  WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are not
&gt; supported
&gt;
&gt; How to solve it from Freeswitch side? I need to offer only RTP/SAVP.
&gt; Also, looking ahead, is it possible to add and modify a=crypto: line, I need
&gt; to send SHA1_80 in offer or both?
&gt;
&gt; Asterisk 1.8.9.2
&gt; FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
&gt;
&gt; SRTP enabled in dialplan/public.xml &lt;action application="export"
&gt; data="nolocal:sip_secure_media=true"/&gt;
&gt;
&gt; Please, advise.
&gt; Thanks!
&gt;
&gt; _________________________________________________________________________
&gt; Professional FreeSWITCH Consulting Services:
&gt; <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
&gt; <a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>
&gt;
&gt; FreeSWITCH-powered IP PBX: The CudaTel Communication Server
&gt; <a href="http://www.cudatel.com">http://www.cudatel.com</a>
&gt;
&gt; Official FreeSWITCH Sites
&gt; <a href="http://www.freeswitch.org">http://www.freeswitch.org</a>
&gt; <a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a>
&gt; <a href="http://www.cluecon.com">http://www.cluecon.com</a>
&gt;
&gt; FreeSWITCH-users mailing list
&gt; <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>
&gt; <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a>
&gt; UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a>
&gt; <a href="http://www.freeswitch.org">http://www.freeswitch.org</a>
&gt;



-- 
Anthony Minessale II

FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a>
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pstn:+19193869900

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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<a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>

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Official FreeSWITCH Sites
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</pre> </blockquote>    

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