is the card sharing irq?
no. this the only card that uses IRQ 30
1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14)
Subsystem: Device 0005:
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop-
ParErr+ Stepping- SERR+ FastB2B- DisINTx-
Status:
I have an mp3 that is 128K, 44.1K stereo.
I convert that to wave 16 bit, stereo, 44.1K
The sound alike at this time.
I want to play them (not just over my sound port) but through asterisk
on select devices/machines that are also running asterisk over the
Console/dsp.
I converted the wave
If you're going to stay with 1.4.X probably g722 would be best for you. If
you work a while with SOX, you should end up with 8K files that sound
almost as good as the 44K wav files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis ge...@pagestation.com wrote:
I converted the wave file to 8K, mono and it doesn't sound very good, I am
also
using 1.4.43 and ulaw,alaw,gsm allowed.
This has been covered just recently, try searching for mp3 on the mailing
list.
What format will
On 11/04/2012 04:17 AM, Andreas Sikkema wrote:
Draytek Vigor2110Vn
Sadly this doesn't seem to do OpenVPN, though it does several other
flavors we might be able to support. Thanks for the tip! Will be
looking into it.
Cheers,
j
--
The Asterisk Development Team has announced the release of Asterisk 11.0.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.0.1 resolves several issues reported by the
community and would have not been possible
and
find the one that sounds the least bad.
--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248
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On 11/4/2012 2:37 PM, Danny Dias wrote:
Thanks Andrew,
But i'm quite confuse with the following:
*Q: Does Digium offer SLA guaranteed support for Asterisk?*
*A:* Yes. Digium offers SLA guaranteed support, to SLA-entitled
customers, for the Certified Asterisk branches. Digium does not offer
On 11/2/2012 4:58 AM, Harish Mandowara wrote:
Hi,
I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi
driver.
Scenario is
The quickest way to get pointed in the right direction would be to
contact Digium support since the issue involves your TDM2400P. Be sure
What is the best way for me to setup Fax Capability with VOIP only.
I have a Asterisk Server hosted on the internet without a modem. I'm
using Flowroute, which is working awesome, for VOIP calls.
I only have a SIP Phone at home and two Printer/Scanner/Fax Printers.
I'm not sure which Fax
On 11/5/12 11:59 AM, Vincent Swart wrote:
You're HDLC error is evident of timing slips.
Use cat /proc/dahdi/1 or 2 or 3
aha.. it does have timing slips...
Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource
CRC4 error count: 6864
E-bit error count: 27603
On 11/05/2012 04:18 PM, Roy Abshire wrote:
What is the best way for me to setup Fax Capability with VOIP only.
You can use T.38, perhaps, if your VoIP provider supports it and you can
get it working. But unless you need faxes to go through the telephony
system (i.e. you have fax machines
Roy,
Many will say that it all depends on your provider supporting T.38, and
that you should forget it otherwise.
My practical experience shows otherwise. I am able to receive faxes on
SIP lines pretty reliably with no T.38 support. The biggest issue for
me is CED tones detection. If CED is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Today I started to experiment with Google Voice and Asterisk-11.0.1.
Following the instructions on the wiki
(https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was
able to make / receive calls quite easily with a single account
Try adding
transport=google-v1 to motif.conf
[google-jd]
context=incoming-motif
disallow=all
allow=speex
allow=ulaw
allow=g722
allow=h264
allow=alaw
*transport=google-v1*
connection=google-jd ; - xmpp.conf
[google-cathy]
context=incoming-motif
disallow=all
allow=speex
allow=ulaw
allow=g722
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit :
Try adding
transport=google-v1 to motif.conf
[google-jd]
context=incoming-motif
disallow=all
allow=speex
allow=ulaw
allow=g722
allow=h264
allow=alaw
*transport=google-v1*
Here are my settings that work. I can make incoming and outgoing calls.
Compare my settings with yours. Also make sure your firewall is open for
port 5222 and 5060 and your RTP port range.
#rtp.conf
[general]
icesupport=yes
rtpstart=15000
rtpend=2
#motif.conf
[default](!)
disallow=all
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