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