Steve Edwards wrote:
On Thu, 17 Apr 2014, Jerry Geis wrote:
I was thinking transcoding was through PRI card - not gsm to ulaw. :)
You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU
I will be using a dell R320 Xeon E5-2420 2G and 4G RAM.
also using a SIP trunk with ulaw/alaw codec.
How many calls could I expect to make at the same time?
no transcoding or anything. Just call a number and play a gsm file.
Thanks,
Jerry
--
On Thu, 17 Apr 2014, Jerry Geis wrote:
I will be using a dell R320 Xeon E5-2420 2G and 4G RAM.also using a SIP
trunk with ulaw/alaw codec.
no transcoding or anything. Just call a number and play a gsm file.
How will you do ulaw - gsm without transcoding?
How many calls could I expect to
On Thu, 17 Apr 2014, Jerry Geis wrote:
I was thinking transcoding was through PRI card - not gsm to ulaw. :)
You can convert the GSM files to ULAW using sox. I tend to transcode
everything to WAV (PCM not that funky 'GSM in WAV') because it is
relatively cheap (CPU cycles) to transcode from
I have been trying to get a feel for scaling or dimensioning using asterisk
11.
if I desire to use something like a dell r320, hardware RAID, 2G E5-2420,
4G RAM
and only SIP trunking using gsm (least bandwidth and no transcoding)
how many calls out can I expect to make at one time and asterisk
On 28/01/14 15:01, Jerry Geis wrote:
I have been trying to get a feel for scaling or dimensioning using
asterisk 11.
if I desire to use something like a dell r320, hardware RAID, 2G
E5-2420, 4G RAM
and only SIP trunking using gsm (least bandwidth and no transcoding)
how many calls out can I
Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk
1.4 + CDRTool with freeradius telephony system.
Asterisk is used only for voice mail and redirectioning calls.
Every calls should pass through mediaproxy so that i can account them.
The goal was to create a
On 7 Jan 2007, at 07:28, Erick Perez wrote:
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot
mini.itx.
Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw
phones +voicemail and *no* call recording?
Yes. I've got the 1ghz version and it is fine (even doing 5
Erick Perez wrote:
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx.
Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw
phones +voicemail and *no* call recording?
Make sure there're no interrupt sharing issues. My old EPIA 1GHz with 2
LAN and 6 USB, had
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx.
Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw
phones +voicemail and *no* call recording?
On 1/5/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Erick Perez wrote:
what if I go with full g711-no
I was thinking of an HP DL140 with two 250gig sata disks and one
3.8Xeon CPU with 2gig RAM.
Should be plenty if not an overkill. One of our setups: 20 phones, 8
outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no
Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a
what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
thanks,
On 1/5/07, Luki [EMAIL PROTECTED]
Erick Perez wrote:
what if I go with full g711-no transcoding?
remember that I will have an E1 coming in, so my usage can be up to 30
channels at once.
if that is an overkill machine config, and for obvious reasons I cant
use old hardware, what are your suggestions?
I would suggest you go for a
Hi,
Some help with dimensioning the server will be gladly accepted.
-50 sip phones (g729) or g711(to avoid transcoding) in LAN
-an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN
-Some sporadic conferencing with no more than 2 sip phones and maybe 2
or 3 calls
Hi Erick -
Some help with dimensioning the server will be gladly accepted.
-50 sip phones (g729) or g711(to avoid transcoding) in LAN
-an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN
-Some sporadic conferencing with no more than 2 sip phones and maybe 2
or 3
On 1/4/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Erick -
Some help with dimensioning the server will be gladly accepted.
-50 sip phones (g729) or g711(to avoid transcoding) in LAN
-an asterisk server (1.4) doing normal pbx functions + voicemail in the same
LAN
-Some sporadic
On Thu, Jan 12, 2006 at 03:05:15PM -0500, Erick Perez wrote:
Hi, is there any good calculator/table/reference about proper dimensioning?
I read the wiki and they basically say xx users run fine in yy hardware
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning.
SO far I read
Hi, is there any good calculator/table/reference about proper dimensioning?
I read the wiki and they basically say xx users run fine in yy hardware
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning.
SO far I read that:
-Run up to 4 E1s per CPU (which one? an i386 or a dual core?
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