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Hi Danny,<br>
<br>
<i>srtpcapable=no</i> in <i>sip.conf</i> didn't work. Asterisk still
wants to establish a SRTP session. <br>
<br>
Regarding your "smart" answer. I disabled the module <font size="3"
color="black" face="Times New Roman"><span style="font-size: 12pt;"><i><span
style="font-style: italic;">res_srtp.so</span></i><span
style="font-style: italic;"> </span><span style="font-style: italic;"></span></span></font>in
<i>modules.conf </i>but even so asterisk still tries to establish the
SRTP session. Because the module is not loaded I get an error and the
call is establish with "normal" RTP. <br>
<br>
<blockquote><i>*CLI> == Using UDPTL CoS mark 5</i><br>
<i> == Using SIP RTP CoS mark 5</i><br>
<i>[Feb 1 09:08:50] ERROR[2225]: chan_sip.c:27972 setup_srtp: No
SRTP module loaded, can't setup SRTP session.</i><br>
<br>
</blockquote>
I want to configure asterisk in a way that it doesn't even tries to
establish SRTP session ... only "normal" RTP.<br>
<br>
Thanks for you help<br>
<br>
- Miguel Baptista <br>
<br>
<br>
On 01.02.2011 18:30, Danny Nicholas wrote:
<blockquote cite="mid:201102011730.p11HUsDR029459@debsweb.debsinc.com"
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<p class="MsoNormal"><b><font size="2" color="black" face="Tahoma"><span
style="font-size: 10pt; font-family: Tahoma; color: windowtext; font-weight: bold;">From:</span></font></b><font
size="2" color="black" face="Tahoma"><span
style="font-size: 10pt; font-family: Tahoma; color: windowtext;">
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b><span
style="font-weight: bold;">On Behalf Of </span></b>Miguel Baptista<br>
<b><span style="font-weight: bold;">Sent:</span></b> Tuesday,
February 01, 2011
11:23 AM<br>
<b><span style="font-weight: bold;">To:</span></b> Asterisk Users
Mailing List -
Non-Commercial Discussion<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re:
[asterisk-users] How
to disable srtp in asterisk 1.8.2.3?</span></font><font color="black"><span
style="color: windowtext;"><o:p></o:p></span></font></p>
</div>
<p class="MsoNormal"><font size="3" color="black"
face="Times New Roman"><span style="font-size: 12pt;"><o:p> </o:p></span></font></p>
<p class="MsoNormal"><font size="3" color="black"
face="Times New Roman"><span style="font-size: 12pt;">Hi again,<br>
<br>
Nobody knows how to disable it? Can at least someone pinpoint me where
can I
check the latest documentation regarding SRTP. Maybe something might
have
change in the meanwhile 'Cause so far it looks like there is a bug in
asterisk.<br>
Well, maybe I should report this bug then.<br>
<br>
- Miguel Baptista<br>
<br>
<br>
On 28.01.2011 18:22, Miguel Baptista wrote: <o:p></o:p></span></font></p>
<p class="MsoNormal"><font size="3" color="black"
face="Times New Roman"><span style="font-size: 12pt;">Hi all,<br>
<br>
I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I
compiled it
with SRTP support. <br>
Everything seems to work OK but I am having a weird issue. I cannot
disable SRTP. I tried the <i><span style="font-style: italic;">encryption=no</span></i>
in <i><span style="font-style: italic;">sip.conf </span></i>and the <i><span
style="font-style: italic;">_SIPSRTP_CRYPTO=disable</span></i> on my
dailplan and
it keeps trying to use the SRTP.<br>
Well, right now I have to have<i><span style="font-style: italic;">
noload=res_srtp.so</span></i>
on my <i><span style="font-style: italic;">modules.conf </span></i>otherwise
I
cannot place SIP calls (cause other ends don't support it)<br>
<br>
Regards,<br>
<br>
Miguel Baptista<br>
<br>
<o:p></o:p></span></font></p>
<pre><font size="2" color="black" face="Courier New"><span
style="font-size: 10pt;"><o:p> </o:p></span></font></pre>
<pre><fieldset class="mimeAttachmentHeader"></fieldset><font size="2"
color="navy" face="Courier New"><span
style="font-size: 10pt; color: navy;">Now that my “smart” answer is out of the way, did you try<o:p></o:p></span></font></pre>
<pre style="margin-left: 0.5in; text-indent: -0.25in;"><!--[if !supportLists]--><font
size="2" color="navy" face="Arial"><span
style="font-size: 10pt; font-family: Arial; color: navy;"><span
style="">-<font size="1" face="Times New Roman"><span
style="font-family: "Times New Roman"; font-style: normal; font-variant: normal; font-weight: normal; font-size: 7pt; line-height: normal; font-size-adjust: none; font-stretch: normal; -x-system-font: none;"> </span></font></span></span></font><!--[endif]-->srtpcapable=no<o:p></o:p></pre>
<pre style="margin-left: 0.5in; text-indent: -0.25in;"><!--[if !supportLists]--><font
size="2" color="navy" face="Arial"><span
style="font-size: 10pt; font-family: Arial; color: navy;"><span
style="">-<font size="1" face="Times New Roman"><span
style="font-family: "Times New Roman"; font-style: normal; font-variant: normal; font-weight: normal; font-size: 7pt; line-height: normal; font-size-adjust: none; font-stretch: normal; -x-system-font: none;"> </span></font></span></span></font><!--[endif]-->in sip.conf?<font
color="navy" face="Arial"><span
style="font-family: Arial; color: navy;"><o:p></o:p></span></font></pre>
<pre><font size="2" color="black" face="Courier New"><span
style="font-size: 10pt;"><o:p> </o:p></span></font></pre>
<pre><font size="2" color="black" face="Courier New"><span
style="font-size: 10pt;">reference: <a moz-do-not-send="true"
href="http://www.voip-info.org/wiki/view/Asterisk+SRTP">http://www.voip-info.org/wiki/view/Asterisk+SRTP</a><o:p></o:p></span></font></pre>
<pre><font size="2" color="black" face="Courier New"><span
style="font-size: 10pt;"><o:p> </o:p></span></font></pre>
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