similar to: Asterisk + G.729 on Sun T1000/T2000

Hello all, First the easy bit, using the info in these mailing lists in respect of HP T750 here is the info for the T1000: Checking device (03F0/1F08) (003/006) FCC no of 5WSUSA-33229-XP-E , suggest it is a tripplite Added the T1000 info to tripplite-hid.c as described recompiled and works fine, thank you. Which driver should I use if I want to connect via the serial port? I am using the

Hi All, We are currently in the process of testing Solaris 10 with ZFS and Oracle and are running it on a T2000. When checking performance statistics on the T2000, we notice that only one thread of the CPU appears to be doing any of the processing. Leaving all other threads seemingly idle. Are there any tuning parameters that need to be set or changed when running Oracle on T2000 using

I have an HP T1000 G3 for looking after three desktop PCs, a switch and a router. The batteries seem to be fine (battery tests, charge levels), but when I'm testing a power-failure, the UPS starts the FSD much too soon (judging by the battery charge level monitored with upsc). The load is about 300W, less than half nominal load, so despite the batteries not being brand new, it ought to keep

I got HP T1000 INTL UPS to work with NUT. Looks like it is more or less equal to HP T750 INTL. I have only tested USB, not serial.

I have made some progress with this. If I add this into my loading script: /etc/init.d/udev stop sleep 1 /etc/init.d/udev start 2>/dev/null sleep 1it works from remastered CD. I have made 25 connection, reboot, reconnect, with no failures. I realise how strange this must sound & I definitely need the sleep commands!I am missing UPS temperature and battery run-time on the

Guru''s, Is anyone shar with me the steps to configure hardware RAID in T2000 server (LSI drivers) and use rootdisk hardware mirror -- Thanks a lot! Malahat Qureshi Ph.D. (MIS) Email: malahatq at hotmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, IHACWHAC (I have a colleague who has a customer - hello, if you''re listening :-) who''s trying to build and test a scenario where he can salvage the data off the (internal ?) disks of a T2000 in case the sysboard and with it the on-board raid controller dies. If I understood correctly, he replaces the motherboard, does some magic to get the raid config back, but even

Here is what I have configured: T2000 with OBP 4.28.6 2008/05/23 12:07 with 2 - 72 GB disks as the root disks OpenSolaris Nevada Build 91 Solaris Express Community Edition snv_91 SPARC Copyright 2008 Sun Microsystems, Inc. All Rights Reserved. Use is subject to license terms. Assembled 03 June 2008

I am running OpenSolaris build 134 on a T1000 (Sparc). It has been an uphill battle getting this far. Many of the commands I find in the various documentation won''t work for me. ------------------------------------------------------------ steve@t1000:~# pfexec pkg install xvm-gui Creating Plan pkg: ''xvm-gui'' matches multiple packages

(Note: resending with proper Subject) If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then

Hi, anyone have any experiences running Sun hardware (maybe a T1000 or 2000) with Ubuntu with Rails? Joe

Hello, All! We was upgrade our Asterisk from version 0.7.2 to 1.0.7. And have big problem. When asterisk starts: ------------------------------------------------------- *CLI> h.323 show codecs Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.723.1{sw} <3> G.711-uLaw-64k{sw} <4> Set: 0: 0: G.729A{sw} <1> G.729{sw}

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com

Dear all I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine. I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications,

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Hi, anyone can confirm if the Linksys's ATA and Router (PAP2-NA and RT31P2-NA) have the same limitation of just one G.729 call like the Cisco ATA 186 ? I'm testing both appliances here and found this issue but could not confirm this anywhere (nothing on the manual, no document or post from any user about this). In my tests they use G.729 only on the first call and G.711 on the

While we're on the subject of G.729... I can end to end use it with no transcoding, but voicemail is the main sticking point for me - I'd need to transcode. So why can't voicemail store the audio in the format it's being streamed in on? Is there a technical reason for no voicemail storage in G.729? We have prompts in G.729, so why not the messages? It doesn't have to mix

Hello, I have trouble activating the G.729 codec in Solaris. It seems the module doesn't try to read the license file at all. Although the module loads (it's listed in show modules), "show g729" gives "No such command" error and this is posted in the messages log: Oct 27 18:09:40 NOTICE[19336] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows "(ULAW)" (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be