RE: [Asterisk-Users] Audio format for announcements
1) Is it possible to store the menu sounds in wav ...sure, just put your 8kHz 16 bit mono files named whatever.wav in /var/lib/asterisk/sounds - asterisk will convert them to what is needed if needed. John This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audio format for announcements
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Sean Adams > Sent: Monday, December 22, 2003 10:50 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Audio format for announcements > > [...] > 2) For my internal SIP phones, I don't care about bandwidth > usage. What > settings will give the best sound quality? Does the protocol (or for > that matter, any particular brand of phones) support uncompressed or > very high bit rate audio for intra-pbx calls? Use g.711ULAW. I belive it is about an 87k uncompressed stream. Sounds better than toll quality to me. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio format for announcements
On Mon, 2003-12-22 at 09:50, Sean Adams wrote: > Hi guys. First off, to the folks at Digium: outstanding work. The fact > that Asterisk is open source puts you right at the cusp of what will be > the most important telecom advance since the transatlantic cable. > > Anyway... a couple newbie questions concerning sound quality - I don't > see any reason why the system should not use the best possible format > for any given connection. > > 1) Is it possible to store the menu sounds in wav/aiff, and let > asterisk compress them to gsm only as necessary? Eg for POTS lines, yes > the lines are crap already, but why butcher the sound any further by > running it through a speech codec? If it is recorded well, and is played on decent interfaces, it won't sound bad. If you so wish, you can store these as wav files in ulaw or alaw format. Look at the codec list and decide what you wish to do. I'm sorry that I have forgotten the name of the person I helped before, but I hosted a prompt for a person so they could see how much of a difference a digital link makes for sound quality. I don't think you would notice a problem no matter what codec within reason if you have a good link. > 2) For my internal SIP phones, I don't care about bandwidth usage. What > settings will give the best sound quality? Does the protocol (or for > that matter, any particular brand of phones) support uncompressed or > very high bit rate audio for intra-pbx calls? All phones should support ulaw or alaw. Those are pretty much the least compressed you will get. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio format for announcements
- Original Message - From: "Sean Adams" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, December 22, 2003 10:50 AM Subject: [Asterisk-Users] Audio format for announcements > > Hi guys. First off, to the folks at Digium: outstanding work. The fact > that Asterisk is open source puts you right at the cusp of what will be > the most important telecom advance since the transatlantic cable. > > Anyway... a couple newbie questions concerning sound quality - I don't > see any reason why the system should not use the best possible format > for any given connection. > > 1) Is it possible to store the menu sounds in wav/aiff, and let > asterisk compress them to gsm only as necessary? Eg for POTS lines, yes > the lines are crap already, but why butcher the sound any further by > running it through a speech codec? > > 2) For my internal SIP phones, I don't care about bandwidth usage. What > settings will give the best sound quality? Does the protocol (or for > that matter, any particular brand of phones) support uncompressed or > very high bit rate audio for intra-pbx calls? > Can't help with the first question, but ulaw/alaw ~= g711, which I think is the biggest eater of bandwidth that I've heard of. See: http://www.voip-info.org/wiki-Codecs - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio format for announcements
Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec? 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? Thanks, Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users