On Mar 17, 2007, at 11:53 AM, Davis Sylvester III wrote:
What do you mean there is no way to do. Others have done this so
there is a way. I thought the taught the purpose of this list was
to HELP
THANKS, for the HELP
Wow, kinda rude don't you think? Matt WAS Helping, why
Performance/Price wise which implementation of the codec is better?
Digium or the Ready Tech/Intel IPP code?
I'm looking at building a 4 PRI g.729 Asterisk box (Dell 2 x dual
core, Digium 4 T1 + echo canceller). Which codec would provide the
best audio quality?
--
Matthew S. Crocker
G.729 is G.729 (assuming same suffixes like B, C, etc.). Audio quality
is exactly the same, or the implementations aren't compatible.
Yes, but depending on the implementation the CPU resources between
two could be quite different. Audio quality could be adversely
affected by inadequate
I'm currently running 1.4b3 with a Digium card and 23 g.729
licenses. Is there a way I can get the g.729 codec work off the CPU
and onto a DSP? Any T1/PRI cards with onboard codec DSPs?
-Matt
--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX
quality is very important.
-Matt
On Oct 20, 2006, at 1:44 PM, Andrew Kohlsmith wrote:
On Friday 20 October 2006 13:01, Matthew Crocker wrote:
I'm currently running 1.4b3 with a Digium card and 23 g.729
licenses. Is there a way I can get the g.729 codec work off the CPU
and onto a DSP? Any T1
Hello,
I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4. I
installed the following
-rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0-
beta2.tar.gz
-rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons-1.4.0-
beta1.tar.gz
-rw-r--r-- 1 root root
On Oct 11, 2006, at 1:25 PM, Matthew Crocker wrote:
Hello,
I'm trying to upgrade an Asterisk 1.2 linux box to Asterisk 1.4.
I installed the following
-rw-r--r-- 1 root root 10908541 Sep 21 13:25 asterisk-1.4.0-
beta2.tar.gz
-rw-r--r-- 1 root root 993921 Sep 21 13:25 asterisk-addons
True but you need to look at the actual speed of a DSL line. a 1.5m/
384k DSL line is actually 1.7m/467k so a chunk of that ATM cell tax
is already factored into the speed you buy from your DSL provider.
If you get an integrated DSL modem/router like a Zoom X5 you can see
the actual
Does anyone have a working sip.conf for a SIP trunk to a Tekelec
T6000 switch. I can get everything to work except the DTMF. The
t6000 requires RFC1833 and I have that in the sip.conf but it still
doesn't seem to work.
Thanks
-Matt
--
Matthew S. Crocker
Vice President
Crocker
, Steve Edwards wrote:
On Thu, 28 Sep 2006, Matthew Crocker wrote:
Does anyone have a working sip.conf for a SIP trunk to a Tekelec
T6000 switch. I can get everything to work except the DTMF. The
t6000 requires RFC1833 and I have that in the sip.conf but it
still doesn't seem to work.
I get
On Sep 5, 2006, at 11:56 AM, Michael Araba wrote:
I have had a bad experience with Asterisk and a Carrier's channel
bank.
The carrier brought in a PRI (data/voice integrated), the data and
voice
channels are split from the channel bank. I connected Asterisk to the
channel bank via T1
On Sep 1, 2006, at 3:36 AM, Jason Lixfeld wrote:
I've been having some problems with a couple of 7970G so I decided
to factory them. In doing so, it seems to have rendered the phones
unbootable as they (as I understand it) are factory'd with some
version of SCCP code, which makes it
I have enabled outside extension '911' and '11' for emergency
service. This way users can either dial '9911' or '911' to get to a
PSAP. I would rather have a couple accidential 911 calls than a
death because someone forgot to dial a 2nd 9. When people are
freaking out they fall back
Add a boolean field to the table then create a view based on the
value of that field
CREATE TABLE `extensions_table_data` (
`id` int(11) NOT NULL auto_increment,
'isActive' boolean NOT NULL default 'True',
`context` varchar(20) NOT NULL default '',
`exten` varchar(20) NOT NULL
Create a context 'suspended' that can *only* dial 911 then move
extensions into that context.
Even if the customer doesn't pay their bill you *still* want them to
be able to dial 911, and probably your customer service number. You
could even create a dialplan in the context to send all
www.zhone.com. Their MALC can handle 500 POTS lines in a 23 shelf
with POTS - VoIP (SIP/MGCP). 'Telco quality' and the per port cost
for high density isn't that bad.
You could probably also go with a bunch of CAC AccesBanks connected
to a CAC Widebank, connected to a Lucent TNT and
Can anyone help point me in the right direction? I have calls coming
into Asterisk over a PRI, all going to the same #. I need to have
asterisk route the calls to a different location based on the NPANXX
of the callerId for the inbound call. Something like
exten = 123,1,$newnumber =
Hello,
How can I build extensions.conf so that Asterisk routes calls based
on the ANI, not the number dialed.
Example:
All calls coming down a PRI are going to the same number. I would
like to route them to a new number based on the Calling-Station-Id.
I.E. All calls from
Hello,
How can I get Asterisk to take an inbound call over a PRI D-Channel,
look up info in a database and send the call out to a new number. I
don't want the patch to occur in the Asterisk box (2 B channels) I
would like to send the call back to the switch on the other end of
the PRI
How about having the asterisk config on CF or USB drives and the OS,
Asterisk on a Linux LiveCD. That way you can mail out PBX upgrades
to the customer they pop it into the CD drive and reboot. Config
for DHCP boot, /etc/asterisk (or /etc entirely) on a USB dongle
would work great.
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
On May 23, 2006, at 1:18 AM, Eric Bishop wrote:
Could someone explain to a non-US dummy the following phrases I
have seen on the list.
I can provide you with tier 1 termination 6/6. I can blend or
NPANXX breakout.
We provide US48 termination, blended rate for 1 MOU and above is .
008
If you use g.711 you should expect the same quality because that is
essentially the same codec the PSTN uses in T1s.
If you use g.729 you should expect some audio degradation as it is a
lossy compression codec. You will lose quality but gain significant
bandwidth reduction
Use a high
Would an internal DUNDi configuration help the asterisk servers share
their extension info? Or, use e.164 with an internal DNS zone to
lookup the routing information. SIP phone logs into Asterisk 'A' and
a script runs to update the e.164 DNS info pointing the DID to
Asterisk 'A'
Take a look at the Net2Net product which was purchased by Paradyne
and then by Zhone (www.zhone.com). They make a unit that will bridge
Ethernet over SDSL lines, 24 pairs will get you 50mbps through the
link. It looks just like ethernet and VoIP will work fine over it.
You can also
Hello,
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like
to be able to direct an inbound fax call into my TNT, have it answer
the fax and send the image file over to Asterisk, or some other
system to deliver to an e-mail address(s). I'm not sure if I need
Asterisk to
On Feb 14, 2006, at 4:33 PM, [EMAIL PROTECTED] wrote:
On Tue, 14 Feb 2006, Matthew Crocker wrote:
I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd
like to be able to direct an inbound fax call into my TNT, have it
answer the fax and send the image file over to Asterisk
Yes, DID on analog line is possible. Your carrier may or may not
support it. For the most reliable service you'll want a 'wink
start' line with DMTF digits. With a DID line you supply
'battery' (-48VDC) to the phone company. They bring the tip to
ground for a second to signal an
I have asterisk 1.0.9 installed with spandsp 0.0.2pre20.
Asterisk crashes on boot while loading app_txfax.so
app_rxfax.so. If I move the files out of /usr/lib/asterisk/
modules asterisk boots fine.
Running on FC3, Linux asterisk.crocker.com 2.6.11-1.27_FC3smp #1
SMP Tue May
1. Is this possible? Can Asterisk serve as a man-in-the-middle
between a
traditional PBX and the telco loops?
Yes, Asterisk can be a PRI client to the telco and a PRI server to
your internal PBX
2. If it's possible, which of the two possibilities above is
better? For us,
rack space
Hello,
I have asterisk 1.0.9 installed with spandsp 0.0.2pre20.
Asterisk crashes on boot while loading app_txfax.so app_rxfax.so.
If I move the files out of /usr/lib/asterisk/modules asterisk boots
fine.
Running on FC3, Linux asterisk.crocker.com 2.6.11-1.27_FC3smp #1 SMP
Tue
Are there any switchvox/fonality type Asterisk based PBXs where I can
buy just the software? I don't want to buy their 'bundles' that come
with junky PC hardware. I just want their software/GUI to run on my
hardware.
Does Asterisk BE come with a GUI management console for managing
Well that would explain the choppy/stuttering sound we get on these
calls
since there is no audio during those error transmissions.
According to Kevin's reply I had my timing logic backwards. Should
I be
using any other timing settings on the Asterisk side?? The tech for
our
legacy PBX
Yes,
Asterisk can do this, you just need to set the zapata.conf for
pri_net instead of pri_cpe. The asterisk server will 'pretend' to
be a switch and the PRI based PBX will never know the difference.
You can also use a Cisco (c2600,2800), CAC (Adit) and Patton to
create a PRI from
I am curious on cpu load, if all dsp functions are done via software
instead of offloaded onto a specialized processor (DSP board) that has
to have some effect on call processing, meaning a more beefy
machine to
handle the load, and the real possiblity of not having a single
board do
Odds are likely that its 1 carrier they use. Without seeing where the
number was ported (requires SS7 access to the carriers involved) its
hard to see which carrier is doing this.
If it really bothers you call broadvoice, and dont bother waiting
in the
queue for techsupprt, go direct :)
All
SS7 is a *signaling* standard, much like SIP, MGCP, H323. It is not
an encoding standard like PCM,g.711, g.729. Asterisk does not
currently support SS7. You could buy a SS7 - SIP (SIP-7) service
from a couple different service providers. Then you could use
Asterisk to take in your IMT
Come June/July an USB/PCI DSP cost effective solution should be
available
to address this issues. It will transcode nearly all codec's.
I am not in position to reveal the company name
at this stage unless MN wants to speak up :)
Put me on the mailing list for the PCI DSP card, I'll beta test if
How are you telling Asterisk to send the call to the fax group? You
should have something in extensions.conf like
exten = _4135551234,1,Dial($FAXTRUNKS/${EXTEN})
Asterisk should send the EXTEN down as a DID to the fax server
-Matt
On Feb 11, 2005, at 11:05 AM, Eric Hall wrote:
Hello Group
I
You can get a support login at CAC for free, You can download all the
manuals in PDF from their support site. www.carrieraccess.com
-Matt
On Jan 29, 2005, at 11:11 AM, Matt Schulte wrote:
We have an old CAC and we're trying to get groundstart working on it,
we
think it may be a dip switch
Yes, that's why I'm posting to this list. I certainly don't want to
have to track my customer's IP addresses. Is there a better way?
Have them go into manual mode and enter your SIP proxy information
directly into the phone, or, have them run their own tftp server and
send them a SIPmac.cnf
On Jan 16, 2005, at 8:45 PM, Joseph wrote:
What would be my best option to receive calls via VOIP.
I would like to use it as an alternative number when my main number is
busy.
The solution is not that easy as in order for customer to be a free
call
DID=Direct Inward Dialing provider would need to
Here is what I am looking for: picture a cisco catalyst switch. (or any
24port rackmount switch thats 1U).
Plug into ports 1-23, a single PRI.
Plug a 100Megabit (or gigabit) ethernet cable into Port 24 that goes
directly to asterisk.
Now you have 1 cable to/from asterisk that can handle all 23
You are one step ahead of me. Everytime the phone tries to load the
SIP 6.3
image, I receive a checksum error and the phone reboots. I have
verified my
sip image is correct ?? Any susgestions ??
Which version of code is the phone currently running? I remember you
had to boot a new 'boot
I gather you've had experience with Rhino.
How does it work with Asterisk?
Does it provide all features? Caller-id, echo cancellation, how's the
voice quality? Transfer? Conferencing? Message indications?
In short, how good is it?
And of course, can you provide configuration samples or point to
Tomoki,
It depends on how many T1 ports you need. You can use Lucent MAX TNT
with VoIP option (available on eBay). You can also us Cisco AS53xx
series equipment (available on ebay)
-Matt
On Dec 5, 2004, at 11:25 PM, tomoki taniguchi wrote:
I am thinking about setting up an asterisk PBX
I just avoid people who think it's ok to create proprietary extensions
to free software. People like that should be ashamed of themselves,
as it's just an insult to the people who have freely contributed to
the project.
I fully agree.
How hard would it be to integrate OpenSS7.org with Asterisk
TC wrote:
I was told that there is $2500 PCI DS3 card available,
It must be a channelized tdm voice ds3
And not just channelized, but channelized down to DS-0. All
channelized cards I've seen only support DS-1 channels.
The MAXIM-921 DS from SBS supports DS3 - DS0 channelized. It is a PMC
for
Here's a question: if the author has purchased a commercial license to
use Asterisk, and I get binary modules from him, I can still use them
with my CVS-based Asterisk, right?
You may be able to do that. You could always run a couple Asterisk
boxes, run IAX2 between them and leave the
Steve,
Who ever is offering the SS7 to * box let me know ASAP. I'll buy it
in a second. I'll beta, I'll work on integration with Verizon East
SS7. Let me know, great timing :)
-Matt
On Nov 16, 2004, at 8:05 PM, Steve Underwood wrote:
Hi Angel,
It is working pretty well. I think it will be
On Nov 1, 2004, at 9:37 PM, Benjamin on Asterisk Mailing Lists wrote:
[EMAIL PROTECTED] wrote:
IAX really isn't the 'one and only' perfect signaling protocol
IAX is *not* a signalling protocol. It is a VoIP protocol.
And that's the whole point. H.323, SIP, et al those are all signalling
protocols,
I'm looking to build an Asterisk system to place in front of my call
center switch. My plan is to eat up 4 PRIs in one office, send to my
other office and convert back to PRI for my legacy switch. Voice
quality is critical, would I be better off going with a Cisco AS5350
using hardware DSPs
On Oct 21, 2004, at 11:39 PM, Kevin P. Fleming wrote:
Matthew Crocker wrote:
I know the AS5400 has STP functionality and I can terminate A-links
from the Verizon STP. What would handle the ISUP messages? Can
Asterisk handle that part of the transaction?
If the AS5400 is doing STP
Is anyone out there using Asterisk to talk SIP with Verisign SIP-7 (SIP
- SS7 gateway service)? I'm looking to control some Cisco AS5400 MGCP
gateways but I need SS7 with Verizon.
Signaling would travel this path:
PSTN - ss7 - Verisign - sip - Asterisk
Bearer traffic would travel this path:
On Oct 21, 2004, at 11:03 PM, Emilio Panighetti wrote:
nd why can't yo make the signaling go directly to the Cisco gateway?
I'm still playing catch up on SS7 so bear with me if I stumble on stuff.
I know the AS5400 has STP functionality and I can terminate A-links
from the Verizon STP. What
What about
[PSTN] -pri- [Asterisk] --T1/ip-- [Asterisk] -pri- PBX
I would need DIDs coming in from the PSTN to be forwarded over a 3mbps
(MLPPP T1) connection to another Asterisk which would regenerate the
PRI back to the PBX.
-Matt
___
Asterisk-Users
On Oct 20, 2004, at 10:42 AM, Kristian Kielhofner wrote:
Michael Di Martino wrote:
I am looking for a loud ringing SIP phone. I am presently using the
Polycom and just cannot loud enough to hear it over the din in a
collocation room.
My Cisco 7960 has the loudest ring that I have ever heard,
I don't know if Cisco officially supports Asterisk but I know they do
provide funding/programmers for many OSS projects (www.vovida.org
VOCAL) being one of them.
-Matt
On Oct 20, 2004, at 1:58 PM, Joseph wrote:
What IP Phones officially support Asterisk. I know that most of them
will work with
Hello,
I'm new to Asterisk, I'll be freeing up some hardware to play with it
next week. Would it be possible to eat up 3 PRIs coming from the
phone company (Lucent 5ESS) into one Asterisk box, ship the traffic
over IP to another Asterisk box and back out as a PRI to a legacy
switch? The
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