search for: baldvin

...nd not found details on the expected frequency range the Speex codec can be expected to carry. Is there any documentation available or a table of some sort that has been compiled which would give an indication of the frequency range based on the various compression options in speex? Best regards, Baldvin Hansson Reykjavik, Iceland baldvin@baldvin.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060531/f6355299/attachment.html

...any interest in some "documentation" text that would help with basic (sometimes silly) questions like the ones I've been asking tonight? Or is such text already available (can't see anything in the PDF for 1.1.9)? I really appreciate you taking time to read and reply. Sincerely, Baldvin > -----Original Message----- > From: Jean-Marc Valin [mailto:Jean-Marc.Valin@USherbrooke.ca] > Sent: 6. j?n? 2005 20:41 > To: baldvin@rogg.is > Subject: Re: [Speex-dev] RTP and jitter buffer relationship > > > > If I would implement my VoIP application using RTP to t...

...d writing back to the socket... not even sure if this is possible. So basically I'm hoping someone has a nice solution for this. With or witout scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or whatever works. I'd really appreciate your input here. Sincerely, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071024/1331c749/attachment.htm

...2) The receiver would receive UDP data and call speex_jitter_put(...) for each packet received. 3) The decoder would call speex_jitter_get(...) to get the next packet to decode and play. Far off, or close to how the big boys would do it? Any information is greatly appreciated. Respectfully, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050606/0db6ff4a/attachment.html

https://bugzilla.samba.org/show_bug.cgi?id=2216 ------- Additional Comments From baldvin@angel.elte.hu 2005-01-06 10:33 ------- I tried the cvs version: it works OK. However, 2.6.3 reproducably hangs. In the NEWS: - Fixed a potential hang when verbosity is high, the client side is the sender, and the file-list is large. OK, maybe this is it. I checked cvs log, and...

...o do this, I'd appreciate your input. It must be able to use a regular analog telephone handset on the far end. I've searched high and low and either I'm not clever enough in using the right terms for this or it is rarely documented? Any details much appreciated. Thank you! Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090526/f70dbd06/attachment.htm

...up at the top of this email), if I knew that there was no data returned from the jitter buffer, I could skip feeding it to the sound card and thus the sound card would be able to "catch up" on the playing of samples from the UDP stream received via the jitter buffer mechanism. Sincerely, Baldvin

...Summary: inplace-if-low-disk Product: rsync Version: 2.6.3 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: baldvin@angel.elte.hu QAContact: rsync-qa@samba.org Maybe it seems a bit perverse, but I have an idea of a feature that I'd really like. It is "--inplace-if-low-disk". Before writing a file, rsync could check the disk space available, and if it is less than the size of the file to...

...sterisk-B. Is it possible to prevent this optimization from happening? Any way to control if it happens at all, or can it be selected on per-call basis somehow? Can I find anywhere more details of call path optimization and it's configuration, use, functionality and behaviour? tnx, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080125/4a0594ff/attachment.htm

...t is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you for your time and effort to respond. Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070609/f6510f5d/attachment.htm

...ven if the monitoring entity needs to be restarted. I'm very curious to hear what your take on this is and if this has perhaps been solved elegantly already? Thank you for considering this question and your time spent thinking about this and possibly replying with your thoughts. Sincerely, Baldvin

...c code but am aparently not bright enough to get the whole point of the short- and long-term margin values etc. Just wonder if it's possible to get a short description of each of these variables, their purpose and how they apply to the whole jitter buffer functionality? Thank you very much. Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20050918/70daef21/attachment.htm

...u may have the correct answer to this scenario: I have an incoming PRI connection to Asterisk 1.4.2. In the office we have two SIP phones and one Zap analog wireless phone. Incoming calls are sent to these three phones in the dial plan extensions.ael with: Dial(Zap/67&SIP/baldvin&SIP/david/${EXTEN}, 50); 1) A new call comes in 2) All phones ring and the first one to pick up the handset gets the call. 3) A new call comes in. 4) The two phones NOT currently busy will ring. 5) The phone answering the first call hangs up. 6) I would not WANT the third phone to also...

...he call established from MD110 to Asterisk (still does not work in the other direction) but no sound is transferred between the two. Just dead silent on both ends. I have some logs and more details if needed and if anyone is ready to listen. Would really appreciate your input on this. tnx, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070609/a438adaa/attachment.htm

...re anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and then and then shining green when there is a call on one of the lines for that port. tnx, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071105/fd39bb60/attachment.htm

..., but I'm just trying to get the hang of this config and I can't find detailed enough documentation for this scenario via usual sources. All information relating to the "correct" or proper configuration of multiple B410P cards in one machine is very much appreciated. tnx, Baldvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071105/9306fd7a/attachment.htm

...Digium cards in one computer (single Asterisk installation)? 2) Do they need to be hard-wired together with a PCM cable like I've seen explained in some beronet manuals (although that was specifically geared towards their cards, I must say)? Thank you for your time and effort! Respectfully, Baldvin

...;s in one machine, four ports in NT mode and four in TE mode. Feeding three ISDN BRI's into the system and three out again: PSTN <-> NT box <-> B410P port in TE mode <-> Asterisk <-> B410P port in NT mode <-> PBX Thank you for your time and effort! Respectfully, Baldvin

...v myfilename.alaw myfinishedfilename.alaw); jump 0001 at dicta-while-recording; } } dicta-while-recording { 0001 => { WaitExten(400); // This is effectively the maximum length of a recording! } } // END OF SAMPLE ************************ Any help is greatly appreciated. Best regards, Baldvin

...y: remote dies, local hangs when disk full Product: rsync Version: 2.6.3 Platform: All OS/Version: Linux Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: baldvin@angel.elte.hu QAContact: rsync-qa@samba.org - I am running rsync through ssh - In a certain phase, two processes exist on both sides - Then, for example when there's a disk full on the remote (target) side, one of the four processes dies, and all the others are hanging on forever (or...