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                <div style="height:1px;">
                </div>I have updated: FreeSWITCH Version 1.0.head (git-fbe4e64 2012-02-22 23-08-19 -0600)<br />
                <br /> After that, in vars.xml:<br /> &lt;X-PRE-PROCESS cmd="set" data="sdp_secure_savp_only=true"/&gt;<br /> in sip_profiles/internal.xml:<br /> &lt;param name="sdp_secure_savp_only" value="true"/&gt;<br /> in dialplan/public.xml:<br /> &lt;action application="bridge" data="{sdp_secure_savp_only=true}sofia/internal/${dialed_extension}%${domain_name}"/&gt;<br /> or<br /> &lt;action application="export" data="nolocal:sdp_secure_savp_only=true"/&gt;<br /> or<br /> &lt;action application="set" data="nolocal:sdp_secure_savp_only=true"/&gt;<br />
                <br /> Debug output return:<br /> EXECUTE sofia/external/123456789@Y.Y.Y.Y bridge({sdp_secure_savp_only=true}sofia/internal/987654321%X.X.X.X)<br /> 2012-02-23 19:47:39.499280 [DEBUG] switch_event.c:1522 Parsing variable [sdp_secure_savp_only]=[true]<br />
                <br /> and two m= lines.<br /> &#160;&#160; v=0<br /> &#160;&#160; o=FreeSWITCH 1329990231 1329990232 IN IP4 X.X.X.X<br /> &#160;&#160; s=FreeSWITCH<br /> &#160;&#160; c=IN IP4 X.X.X.X<br /> &#160;&#160; t=0 0<br /> &#160;&#160; m=audio 29028 RTP/SAVP 0 8 101<br /> &#160;&#160; a=rtpmap:101 telephone-event/8000<br /> &#160;&#160; a=fmtp:101 0-16<br /> &#160;&#160; a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:u6q91ZUqwrEMnd8/w3YsLijAMYfvCoPc/63vJL5o<br /> &#160;&#160; a=silenceSupp:off - - - -<br /> &#160;&#160; a=ptime:20<br /> &#160;&#160; m=audio 29028 RTP/AVP 0 8 101<br /> &#160;&#160; a=rtpmap:101 telephone-event/8000<br /> &#160;&#160; a=fmtp:101 0-16<br /> &#160;&#160; a=silenceSupp:off - - - -<br /> &#160;&#160; a=ptime:20<br />
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                <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: 23 февраля 2012, 04:41:46<br /> Тема: Re: [Freeswitch-users] How to disable two 'm=' lines<br /> </div>
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                <blockquote class="ukr_editor_quotation" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><pre>
i missed the bit that it was related to srtp

this is a joke that they don't accept both its part of the specs.

anyway,  Update to HEAD and try this new variable (i had to resist
calling it cmon_asterisk_really=true)

sdp_secure_savp_only=true




2012/2/22 Mihail &lt;<a href="mailto:pazuzu@ukr.net">pazuzu@ukr.net</a>&gt;:
&gt; Thank you for replay.
&gt; I tried to set up in ../conf/vars.xml
&gt;
&gt;
&gt;
&gt;
&gt; &lt;X-PRE-PROCESS cmd="set" data="sdp_m_per_ptime=false"/&gt;
&gt;
&gt; After reloadxml and shutdown.
&gt; It did not help. An issue still exist.
&gt;
&gt; --- Исходное сообщение ---
&gt; От кого: "Anthony Minessale" &lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt;
&gt; Кому: "FreeSWITCH Users Help" &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>&gt;
&gt; Дата: 21 февраля 2012, 18:29:21
&gt; Тема: Re: [Freeswitch-users] How to disable two 'm=' lines
&gt;
&gt;
&gt;
&gt; set the global var sdp_m_per_ptime=false in vars.xml
&gt;
&gt;
&gt; 2012/2/21 Mihail &lt;<a href="mailto:pazuzu@ukr.net">pazuzu@ukr.net</a>&gt;:
&gt;&gt;
&gt;&gt; Hello.
&gt;&gt;
&gt;&gt; I try to connect Asterisk as a client to Freeswitch using TLS and SRTP.
&gt;&gt; Outbound direction (from Asterisk) - TLS/SRTP works fine.
&gt;&gt; There is an SRTP issue with inbound calls.
&gt;&gt; FS in SDP offer sends two 'm=' lines:
&gt;&gt;
&gt;&gt;    m=audio 23036 RTP/SAVP 0 8 101 13
&gt;&gt;    a=rtpmap:101 telephone-event/8000
&gt;&gt;    a=fmtp:101 0-16
&gt;&gt;    a=crypto:1 AES_CM_128_HMAC_SHA1_32
&gt;&gt; inline:bdDpQbqc5iaGvVhCilOd5nXOKHLfGGm8J3mjsLIa
&gt;&gt;    a=ptime:20
&gt;&gt;    m=audio 23036 RTP/AVP 0 8 101 13
&gt;&gt;    a=rtpmap:101 telephone-event/8000
&gt;&gt;    a=fmtp:101 0-16
&gt;&gt;    a=ptime:20
&gt;&gt;
&gt;&gt; Receiving that Asterisk reply:
&gt;&gt;  WARNING[9480]: chan_sip.c:8847 process_sdp: Multiple audio streams are
&gt;&gt; not
&gt;&gt; supported
&gt;&gt;
&gt;&gt; How to solve it from Freeswitch side? I need to offer only RTP/SAVP.
&gt;&gt; Also, looking ahead, is it possible to add and modify a=crypto: line, I
&gt;&gt; need
&gt;&gt; to send SHA1_80 in offer or both?
&gt;&gt;
&gt;&gt; Asterisk 1.8.9.2
&gt;&gt; FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)
&gt;&gt;
&gt;&gt; SRTP enabled in dialplan/public.xml &lt;action application="export"
&gt;&gt; data="nolocal:sip_secure_media=true"/&gt;
&gt;&gt;
&gt;&gt; Please, advise.
&gt;&gt; Thanks!
&gt;&gt;
&gt;&gt; _________________________________________________________________________
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&gt;&gt;
&gt;
&gt;
&gt;
&gt; --
&gt; Anthony Minessale II
&gt;
&gt; FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a>
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&gt;
&gt; AIM: anthm
&gt; MSN:<a href="mailto:anthony_minessale@hotmail.com">anthony_minessale@hotmail.com</a>
&gt; GTALK/JABBER/PAYPAL:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>
&gt; IRC: irc.freenode.net #freeswitch
&gt;
&gt; FreeSWITCH Developer Conference
&gt; sip:<a href="mailto:888@conference.freeswitch.org">888@conference.freeswitch.org</a>
&gt; googletalk:<a href="mailto:conf+888@conference.freeswitch.org">conf+888@conference.freeswitch.org</a>
&gt; pstn:+19193869900
&gt;
&gt; _________________________________________________________________________
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&gt;
&gt;
&gt; _________________________________________________________________________
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&gt;
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&gt;
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&gt;
&gt; FreeSWITCH-users mailing list
&gt; <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>
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&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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AIM: anthm
MSN:<a href="mailto:anthony_minessale@hotmail.com">anthony_minessale@hotmail.com</a>
GTALK/JABBER/PAYPAL:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:<a href="mailto:888@conference.freeswitch.org">888@conference.freeswitch.org</a>
googletalk:<a href="mailto:conf+888@conference.freeswitch.org">conf+888@conference.freeswitch.org</a>
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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</pre> </blockquote>    

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