similar to: Multi-bitrate codecs

Hi, I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as sip-to-pstn GW. The issue is that when a call comes in from the pstn, asterisk correctly contacts the router, which in turns send a "183 Session progress". Obviously, asterisk thinks that the telephone is not ringing (because it expects a "180 Ringing") and we have no ringback on the pstn side. Putting a

Hi, I've got a problem with some grandstram devices (namely a couple of budgetone 101 and an ata-486). The point is that, unless I use inband for DTMF, asterisk ignore the first digit dialed. Inband DTMF forces me to use A-law/Mu-law, which is not what I want. BTW, this appens after a Playtones(), waiting for user entering an extension. I've tried many solutions, played around with

codeweavers-wine-20010112-1 hung during install (rpm) at the creation of KDE/Gnome menus; since I don't use either, I thought, no prob and ran winesetup from Blackbox. I skipped re-making the links on starting winesetup and went right for the configuration. which also failed. This is the message it generated: Error: Invoking 'wish Main.itcl' in /opt/wine/bin/winesetuptk

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I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? TIA, Simone.

I'm not an expert either, but I see people choosing iLBC over speex all the time with asterisk; partly it's because they have more market share in hardphones, and partly it's because of marketing and such. (another reason is that iLBC source is included in asterisk, and speex is only compiled in if you have the speex development stuff on your machine when you compile

I think it is a important bug. When there are mutiple write to a xapian db, it should raise db locked error. But I found the DB is currupted after that. It happened very often in our system. Yes, I know xapian support only one write. But admin mistake may happen. Can this be solved? thanks. -- Pan

Hello developers, I have a question about supported bitrates for SPEEX. Regardless of whether I specify a 10kbps or a 8kbps bitrate, the encoded .spx file is the same size and have the same PESQ value. This makes me wonder if SPEEX supports arbitrary bitrates, or only a certain set. A few more details about my configuration: I am using SPEEX encoder version 1.1.12 I am encoding a narrowband

We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports.

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure

Greetings all, I'm trying to improve the codec selection on a few of the asterisk boxes we have to keep the g729 licences free for calls from ATAs that don't support anything apart from g711 and g729. GSM seems to offer noticably inferior call quality (at least when using a softphone + decent headphones), but it's about where I want the bitrate to be. I know there are lots of Speex

Hello: How do i make a mutiple select list for "nomodel" objects?I made that by using something like:<select name="" multiple="mutiple"><%=option groups collection for select(....)-%></select>.But when i select mutiple items and submit them,I can only get one selected item? Who can help me? -- Posted via http://www.ruby-forum.com/.

Hi, I am looking how to setup speex codec in codecs.conf for 8 Kbps and 6 Kbps. In config file are many parameters for setting. I don't know what is need to change for narrowbad like 8 Kbps and 6 kbps. Any suggestion? [speex] ;0-10 quality => 4 ;0-10 complexity => 4 ; true / false enhancement => true ; true / false vad => false ; true / false vbr => false ; 0 = off, otherwise,

Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, Linphone just sends raw packets, as specified in the RTP draft. Jean-Marc Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > jonathan blais wrote: > > I'm using Linphone. I tested with Asterisk and Speex only, I created > > a channel with echo and it worked. It seems to have problem when >

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis/attachments/20041014/25f999a2/attachment.htm -------------- next part -------------- ? dear All, Iam working on OGG VORBIS TREMOR code to port and optimize on ARM processor. Can any one make me clear regarding the relation ship between this 1. What is the relation ship between Quality factor and Bitrates?

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

hi everybody, how I do to set a mutiple virtual host on httpd configuration? I need to host multiple sites with the same IP address what must be the httpd.conf configuration file and into /etc/httpd/conf.d/? thanks in advance kind regards, ./p

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

dear vorbis developers, it would be very useful, if oggenc would support constant low-bandwith bitrates for testing. something from 20kbps to 44kbps, so anyone could test streaming over modem and isdn. audio quality doesn't matter. mörk --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

Hi folks, Ok, I've seen this question go unanswered on the mailing list, and I assume it's because no one had the heart to break the bad news to the guy asking, but be honest with me, I can take it. At this time it's flat impossible to have multiple IAX phones behind a NAT without using an * gateway because there's no way to have a client listen on a port besides 4569. Is