Hi,<div>you need to build Asterisk with SRTP support...</div><div><br></div><div><b style="font-family:'Times New Roman';font-size:medium;font-weight:normal"><span style="font-size:15px;font-family:Calibri;font-style:italic;vertical-align:baseline;white-space:pre-wrap">wget <a href="http://sourceforge.net/projects/srtp/files/latest/download" target="_blank">http://sourceforge.net/projects/srtp/files/latest/download</a></span><span style="font-size:15px;font-family:Calibri;vertical-align:baseline;white-space:pre-wrap"> -O srtp-latest.tgz</span><span style="font-size:15px;font-family:Calibri;font-style:italic;vertical-align:baseline;white-space:pre-wrap"></span><br>
<span style="font-size:15px;font-family:Calibri;font-style:italic;vertical-align:baseline;white-space:pre-wrap">tar -zxvf srtp-latest.tgz</span><br><span style="font-size:15px;font-family:Calibri;font-style:italic;vertical-align:baseline;white-space:pre-wrap">./configure --prefix=/libsrtp</span><br>
<span style="font-size:15px;font-family:Calibri;font-style:italic;vertical-align:baseline;white-space:pre-wrap">make && make install</span></b></div><div><br></div><div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i>And for Asterisk...</i></span></font></div>
<div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i>./configure --with-srtp=/libsrtp</i></span></font></div><div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i><br></i></span></font></div>
<div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i>this should work...</i></span></font></div><div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i><br></i></span></font></div>
<div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i>I did some changes to the sipml5 client and wanted to share this with you guys... Actually only 2 simple changes...</i></span></font></div><div>
<a href="https://github.com/mailsvb/sipml5" target="_blank">https://github.com/mailsvb/sipml5</a>
</div><div><br></div><div><font face="Calibri"><span style="font-size:15px;white-space:pre-wrap"><i>- The main config section has been splitted and made a little more flexible, see </i></span></font><a href="http://i45.tinypic.com/10x59o7.png" target="_blank">http://i45.tinypic.com/10x59o7.png</a></div>
<div>- Main call.html file has been renamed to .php and some code has been added that will replace the "something.invalid" with the actual IP of your client PC.</div><div><br></div><div>Currently I am able to register and at least make my softphone ring ;-) As soon as I answer the outgoing call from sipml5 in the softclient, I get an error in sipml5...</div>
<div><br></div><div>You can find my console output here <a href="http://pastebin.com/jdkXSMSD">http://pastebin.com/jdkXSMSD</a> </div><div>I will continue investigating tomorrow...</div><div><br></div><div>best regards,</div>
<div>Sven</div><div><br><div class="gmail_quote">2012/8/20 Juan Castro <span dir="ltr"><<a href="mailto:jcastro@instant.com.br" target="_blank">jcastro@instant.com.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Put my sipml5 changes there. By the way, this is what happens when I<br>
try to call a X-Lite extension from a sipml5 extension:<br>
<br>
jcvmasterisk1*CLI><br>
== Using SIP RTP CoS mark 5<br>
[Aug 20 17:24:02] ERROR[22737][C-00000009]: chan_sip.c:32140<br>
setup_srtp: No SRTP module loaded, can't setup SRTP session.<br>
[Aug 20 17:24:02] WARNING[22737][C-00000009]: chan_sip.c:9974<br>
process_sdp: Can't provide secure audio requested in SDP offer<br>
jcvmasterisk1*CLI><br>
<br>
Trying to do the reverse... X-Lite stays in "Calling..." - in sipml5,<br>
the right pane, with the local webcam thumbnailm, pops up, but no<br>
"Answer" button. Only "Call" and "Hangup". Also, after a loooong time,<br>
I get a ringing tone in X-Lite. And the webcam thing never goes away<br>
in sipml5. What I get in the log is just this:<br>
<br>
jcvmasterisk1*CLI><br>
== Using SIP RTP CoS mark 5<br>
-- Executing [2010@demo:1] Dial("SIP/2012-00000004", "SIP/2010")<br>
in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called SIP/2010<br>
jcvmasterisk1*CLI><br>
<br>
sipml5 to sipml5: "Not acceptable here". And the destination extension<br>
is totally inert. Log:<br>
<br>
jcvmasterisk1*CLI><br>
== Using SIP RTP CoS mark 5<br>
[Aug 20 17:30:58] ERROR[22747][C-0000000c]: chan_sip.c:32140<br>
setup_srtp: No SRTP module loaded, can't setup SRTP session.<br>
[Aug 20 17:30:58] WARNING[22747][C-0000000c]: chan_sip.c:9974<br>
process_sdp: Can't provide secure audio requested in SDP offer<br>
jcvmasterisk1*CLI><br>
<br>
Meh, same thing as simpl5-to-plain-SIP.<br>
<span><font color="#888888"><br>
Juan<br>
</font></span><div><div><br>
On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham <<a href="mailto:lathama@gmail.com" target="_blank">lathama@gmail.com</a>> wrote:<br>
> On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro <<a href="mailto:jcastro@instant.com.br" target="_blank">jcastro@instant.com.br</a>> wrote:<br>
>> Hoo-hah. It registers. Progress!<br>
>><br>
>> Now... media. Or not.<br>
>><br>
>> On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp <<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>> wrote:<br>
>>> ----- Original Message -----<br>
>>>> ><br>
>>>> > The complete URL to use is http://<asterisk IP address or<br>
>>>> > host>:8088/ws<br>
>>>> ><br>
>>>> > Note the /ws at the end. WebSocket support is only available there.<br>
>>>> > Doing otherwise would have required core HTTP server changes,<br>
>>>> > which I wanted to avoid. Depending on what you are testing with<br>
>>>> > you may need to change it slightly to add that in.<br>
>>>><br>
>>>> Well, I did the following changes in sipml5 and now I get a "Bad<br>
>>>> Request" on REGISTER, instead of 404. Clearly, I'm still missing<br>
>>>> something. Here are the changes I made:<br>
>>><br>
>>> You are probably getting hit by a bug in Asterisk 11 that has been fixed.<br>
>>><br>
>>> It's noted here in the wiki page I'm working on: <a href="https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support" target="_blank">https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support</a> along with a work around via configuration.<br>
>>><br>
>>> --<br>
>>> Joshua Colp<br>
>>> Digium, Inc. | Software Developer<br>
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
>>> Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
>>><br>
>>> --<br>
>>> _____________________________________________________________________<br>
>>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
>>> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
>>><br>
>>> asterisk-users mailing list<br>
>>> To UNSUBSCRIBE or update options visit:<br>
>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
>><br>
>><br>
>><br>
>> --<br>
>> Juan Carlos Castro y Castro<br>
>> Instant Solutions - Telefonia Gerando Resultado<br>
>> <a href="http://www.instant.com.br" target="_blank">http://www.instant.com.br</a><br>
>> Principais capitais: 4063-6100<br>
>> Demais regiões: (11)4063-6100<br>
>><br>
>> --<br>
><br>
> Juan<br>
><br>
> Matt just opened<br>
> <a href="https://issues.asterisk.org/jira/browse/ASTERISK-20267" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-20267</a> to document<br>
> some of this. Feel free to pipe in.<br>
><br>
> --<br>
> ~ Andrew "lathama" Latham <a href="mailto:lathama@gmail.com" target="_blank">lathama@gmail.com</a> <a href="http://lathama.net" target="_blank">http://lathama.net</a> ~<br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
<br>
<br>
--<br>
Juan Carlos Castro y Castro<br>
Instant Solutions - Telefonia Gerando Resultado<br>
<a href="http://www.instant.com.br" target="_blank">http://www.instant.com.br</a><br>
Principais capitais: 4063-6100<br>
Demais regiões: (11)4063-6100<br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br></div>