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...g it from R. (I am also new to shared libraries...) My problem is that my C code uses a C function contained in another library (".a", not ".so" - is that the problem?). This is how I compile the file with the functions I want to call from R: cd /export/home/gpetris/ R SHLIB tryit.c cc -I/export/home/share/R/R-1.4.0/lib/R/include "-I/export/home/gpetris/C" -I/usr/local/include -I/export/home/share/include -KPIC -xO5 -dalign -xlic_lib=sunperf -c tryit.c -o tryit.o cc -G -o tryit.so tryit.o "-L/export/home/gpetris/lib/SUN/32bit -lutil -lrand" -L/usr/loc...

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ <http://tryit.jssip.net/> <http://tryit.jssip.net/ <http://tryit.jssip.net/>> is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) afte...

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ <http://tryit.jssip.net/> is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) after marking video packet I?m able to make Dial() between two webrt...

...May 12 14:33 .davi/ -rw-rw-r-- 1 dovecot Everyone 0 May 12 13:46 dovecot-shared drwxrwS--- 2 dovecot Everyone 4.1k May 12 13:48 new/ -rw------- 1 mark Everyone 6 May 12 14:07 subscriptions drwxrwS--- 2 dovecot Everyone 4.1k May 12 13:48 tmp/ drwx------ 5 mark Everyone 4.1k May 12 14:30 .tryit/ The directories .davi/ and .tryit/ were created by users david and mark respectively. These are real system users. They are both members of the Everyone group. Any suggestions appreciated. Thanks, Mark

...VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best } --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070913/5d62ee74/attachment.htm

Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.<div><br></div><div>I configure my Asterisk 11.7.0 to work wit WEBRTC.</div><div><br></div><div>Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it.</div><div><br></div><div>here is the part of the SIP DEBUG</div><div><br></div><div><--- SIP read from WS...

Dear R helpers, I try to use the 'delaunayn' function in the 'geometry' package for Delaunay triangulation in 2 dimensions. For the four following points, I get a warning message : > coord=matrix(ncol=2,byrow=TRUE,c(622633,7073452, + 621228,7073517, + 621879,7071762, +

...> Jean-Marc > > > > --------------------------------- Yahoo! Mail is the world's > favourite email. Don't settle for less, sign up for your freeaccount > today. --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20070722/5624f30b/attachment.htm

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) after marking video packet I?m able to make Dial() between two webrtc peers but i get one way audio...

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) after marking video packet I?m able to make Dial() between two webrtc peers but i get one way audio...

...stall setup when used with Asterisk+freepbx+Sangoma. Also how do I enable DTMF hardware detection. -- Deepak Linux your Life, Don't Window it [[]] { All for the best } --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070721/cafb44e2/attachment.htm

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...se it proves that Asterisk doesn't have to be exposed as the HTTP WebSocket server: all the WebSocket handshake and message parsing is done by the proxy. Specific versions tested: - Asterisk 11.4 built from SRPM on CentOS 6 + EPEL6 - repro 1.9.0~alpha0 package from Debian experimental - JsSIP `tryit' client - Google Chrome Just some more notes about problems encountered with the Asterisk SRPM: it doesn't seem to know anything about /usr/share/asterisk/sounds - even though I install both the gsm and ulaw sounds RPMs, it always gives errors such as file.c:701 ast_openstream_full: File d...

...his, you ask? Simple. I want to put bulletin messages, reminder messages, corporate communication snippets, etc for employees to hear - no more than 3 seconds. Thanks for your suggestions. Jeng --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070806/0da57287/attachment.htm

...n => 555,1,Macro(exten-vm,555,555) exten => 555,n,Hangup exten => 555,hint,SIP/555 exten => ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL) exten => ${VM_PREFIX}555,n,Hangup -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070625/2583b643/attachment.htm

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). i have two questions and i hope you could give me some advise. 1) after marking video packet I?m able to make Dial() between two webrtc peers but i get one way audio...

...sues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now I?m using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything work perfect on echo test :). > > i have two questions and i hope you could give me some advise. > > 1) after marking video packet I?m able to make Dial() between two webrtc peers but i g...

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details

...or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who knows. Tryit now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/cac8522c/attachment.htm