search for: masakazu

...N}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename. but ${EXTEN} meaning 'i' that isn't dialed number. Does anyone have good idea? please help --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan. http://www.dairiten.com:81/modules/news/ powered by xoops at http://www.xoops.org

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Hi Masakazu, I have reproduced the Speex high frequency noise issue on a Blackfin 537 by sending 'silence' into the Speex fixed point encoder. By recording the Speex decoder output and plotting the spectrum using Audacity the following two frequencies. 1596Hz at -31.3dB 3200Hz at -48dB These two...

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

...N}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename. but ${EXTEN} meaning 'i' that isn't dialed number. Does anyone have good idea? please help --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan. http://www.dairiten.com:81/modules/news/ powered by xoops at http://www.xoops.org _______________________________________________ Asterisk-Users mailing list mailto:Asterisk-Users at lists.digium.com http://lists.digium...

....0.1:5060;branch=76bf5f20 >From: <sip:110 at 192.168.0.1;user=phone>;tag=4dad5671 To: "mack2" <sip:mack2 at 192.168.0.1>;tag=4wvwq4r7lt Call-ID: 3e6096da795e-fc52831snxub at 210.194.204.16 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.14:5060 --- Masakazu Nakano as mack at irc http://www.dairiten.com:81/

I am running Asterisk CVS-04/16/03-18:57:13, and mpg123-0.59r It all sounds great and it plays at the correct pitch and speed. However at the end of the file it simply does nothing. It does not go on the the next step in the extension.conf nor does it hang up. It just sits there. During play I have two processes running for the mp3 stream: root 6300 6299 8 22:32 ?

...ble-fixed-point --disable-float-api --disable-vbr Cheers! Jason Jean-Marc Valin wrote: > What version did you use. Some older releases had encoder-side issues > with some times, but that should be fixed now. > > Jean-Marc > > Jason Hennigar a ?crit : > >> Hi Masakazu, >> >> I have reproduced the Speex high frequency noise issue on a Blackfin 537 >> by sending 'silence' into the Speex fixed point encoder. By recording >> the Speex decoder output and plotting the spectrum using Audacity the >> following two frequencies. &g...

Hi Isamar maybe I think disclose your code to CVS is best and fast :-) mack > > Hi folks, > > I'm trying now to play with fsk_modem.c and callerid.c > to get the Japanese callerid working and I already got to make some > steps.. > I don't know if anybody accomplished that already... but > Since two or more minds think better than one, send private messages >

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

Hi, speex developers! I'm new to speex, and trying to build a simple echo application. something like this. microphone -> speex encoder -> speex decoder -> speaker I first test it the simple way, microphone -> do nothing -> speaker and this works fine. But when I try to encode speex, and decode it, it works, but some high frequency noise is added. How can I fix it? The

Hi, speex developers! I'm new to speex, and trying to build a simple echo application. something like this. microphone -> speex encoder -> speex decoder -> speaker I first test it the simple way, microphone -> do nothing -> speaker and this works fine. But when I try to encode speex, and decode it, it works, but some high frequency noise is added. How can I fix it? The

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn