similar to: G722?

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

Hallo, I use dovecot 2.1.7 with mdbox format and I specified a separate location for mailbox index files with mail_location = mdbox:/dovecot/storage%h:INDEX=/dovecot/index%h With this setting I get for every mailbox a directory under both filesystems /dovecot/index and /dovecot/storage, e.g. for the inbox it looks like that: # ls -lR /dovecot/index/home/hrz/myuser/mailboxes/INBOX

Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael

Has anyone used the Polycom HDvoice phone yet? I am curious if it uses a different codec. Does it actually sound any better?

Has anyone done any integration work with Covad's hosted solution ? I am considering Covad's hosted solution and want to be able to use Asterisk to develop some other apps. Anyone else tried this ? how did Covad react. I know they use MGCP. Another thing, the Cisco reseller rep tells me if I have a bunch of 7960's setup for MGCP (for use with Covad) I will need to get these

So here's a straw-man proposal for a github migration: 1. Register an official github project with the llvm foundation. 2. Setup another (read-only) mirror of llvm.org/git at this github project 3. Make sure we have ala llvm-project-submodules setup in the official account. (Optional or necessary for the buildbots?) 4. Update the buildbots to pick up updates and commits from the official git

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

I may have need for a small multi-port FXO i/f. Anyone have hands-on experience with this linksys device? On past projects I've used SPA-3000 & Digium TDM400 and found them unsatisfactory. Michael

On this weeks VUC call we will collectively be our own guests. That is, we'd like to know what was the big issue that impacted YOU in 2010? All opinions welcome. Here are a few things to get you thinking in advance: - Apple's Antenna-gate - Asterisk 1.8 Launches - Amazon EC2 as a DOS platform - Cisco launched UMI video conference device - More HDVoice capable phones - Skype Outage - VoIP

Does anyone know if support for the Intel 875P/ICH5 motherboard chipset has yet made it into the stable branch? (Is release 4.9 likely to have it?) I am mainly interested in the IDE and "native" serial ATA devices. There is also a new Intel ethernet controller chip, 82547EI, that is designed to interface directly with the 875P chip. The currently supported chip list only goes up to

OK. Now I show my ignorance. What's the difference between a T-1 and a channelised T-1? I see that Covad's voip service (formerly GoBeam) requires a channelised T-1. Then I read recently on the list that many T-1s being installed are actually HDSL....which would be not a T-1 at all...right? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris

A few hours ago I downloaded .../i386/ISO-IMAGES/4.9-RC1-i386-disc1.iso and made kern/mfsroot floppies from it. If I enable my ICH5R SATA controller in "native" mode, the 4.9-RC1 GENERIC kernel hangs solidly (only the system reset button can unwedge it) during device configuration. The last line of bootstrap monologue written by the kernel is: plip0" <PLIP network

I thought XO was reselling Level 3s (old Genuity assets) network/voip just like Qwest ? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jk@bingoconsulting.com Sent: Wednesday, October 11, 2006 3:38 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Load balance Asterisk server,when it is a SIP

Hi, spurred by a severe mistunderstanding on my part, I found myself in a very OSI-8/PEBCAC situation. I would like to politely suggest two hopefully small alterations to make such misunderstandings more difficult and stop a misguided user much earlier. Please bear with my explanation, I try to be brief. My problem's root cause was that misunderstood what a ssh certificate actually is.

Does anyone have an actual delivery date on the new Polycom HD IP650's? I'm getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061018/9937184f/attachment.htm

In case there are any Debian developers around here, I wanted to point out message <20020828154322.GA15114@chulak.naquadah.org> on the debian-devel list this morning. Another Debian developer is proposing to submit an mp3->vorbis transcoding program for inclusion in Debian. I have objected to this on the grounds that the resulting vorbis files will sound like crap, and I have also pointed

Hi, I wanted to follow up on this thread about WB support on the MeetMe bridge that is in 1.6.2. Does it only work for G722 or any WB codec ? I am working with another 16k WB codec that I can transcode to 722 and vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB codec natively(without downscaling). Thanks, Serhad Doken ------------------> Razza wrote:

I apoligize in advance for this newbie question on what I perceive as a mostly advanced level list... I did some searching, but would like some of your expert opinions. I'm building an asterisk server to be used in the home, both to learn, and as proof of concept of applying this solution in a home. To keep costs down, I'm considering one X100P ($25 ebay clone) card to connect the