.conf, nor what device to point pppd to, in order to do
this.
Sincerely,
Rusty Dekema
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were configured:
https://gist.github.com/DrDeke/9436874
Running dahdi_tool shows only one span; "DAHDI_DUMMY/1 (source: HRtimer)",
in the "UNCONFIGURED" state.
If anyone has any ideas for me to try, or would like to see any further
debugging
It's normal to have to wait (under a second in your case) for a dial
tone from the phone company when seizing a line.
If you were placing a call on a phone directly connected to the phone
company, the time it takes to physically pick up the phone and move
your hand to the dial normally takes at l
On 6/22/06, BerkHolz, Steven <[EMAIL PROTECTED]> wrote:
I am thinking of getting an asterisk user group together for either SE
Michigan or just Metro-Detroit.
I'm in Ann Arbor and would be interested in such a group; if you
create a mailing list for it, could you please add me?
Thanks,
Rusty
_
In Europe, the 900 and 1800MHz bands are used for GSM. In the USA, the
800 (or "850" as some call it) and 1900MHz bands are used for GSM as
well as other protocols.
T-Mobile USA uses 1900MHz GSM exclusively, although they do have a few
territories in which GSM 800/850 roaming is allowed. So if yo
Hi,
Does anyone know of a softphone that supports G.722; preferably one
that is available free of charge? Either IAX2 or SIP would be fine.
Thanks,
Rusty
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On 4/25/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Unless you have a top of the line Pocket PC don't even bother. Most
> inexpensive units like the T-Mobile MDA just don't have the processing power
> to handle VoIP. I have tried ESJPhone, SJPhone, and some other one which I
> forgot about alre
On 4/25/06, Sean Cook <[EMAIL PROTECTED]> wrote:
> This worked perfectly! Thank you!
>
> Sean
Now, I think the question is, does your ATA actually support
rotary/pulse dialing? Mine (SPA-2000) did not. I bought a (very cheap)
MITEL-1 "Smart Dialer" and went through a RIDICULOUS amount of pain
tryi
The "PC" port on a BT-102 should work with any computer that has an
Ethernet card. Have you tried these phones with other computers than
the Mac Minis you mention? It shouldn't make any difference whether
the computer is a Mac, PC or anything else. Perhaps something is wrong
with the BT-102s you ha
On 4/14/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
> I believe the TDM2400 has the capability of doing on-card fxo->fxs data
> flows (without hitting the pci bus), but that function has not yet been
> implemented. Its basically "required" to support faxes in an analog
> environment. When it is imp
Well, I guess that would depend on what you mean by multi-layered
access control. Kind of like your subject line asks.
-Rusty
On 4/13/06, Carey Mould <[EMAIL PROTECTED]> wrote:
> How does multi-layered access control work in asterisk?
> ___
> --Bandwidt
On 4/11/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> In the US, bri & pri's are less popular for lots of reasons, part of
> which is the cost of implementing the necessary software on the CO
> switch. Siemens (as one example only) charges their small CO customers
> $7,000 to implement the softwa
On 4/11/06, Andy Tan <[EMAIL PROTECTED]> wrote:
> Hi,
>
> understand that the bandwidth utilized for each call is dependent on the
> codec used, wonder if Asterisk can monitor the total bandwidth utilized
> and restrict/reject new calls when the resource is insufficient to
> support them reliably?
On 4/4/06, Anton Krall <[EMAIL PROTECTED]> wrote:
> It really makes that much diff. using slinear vs. ulaw?
I wouldn't think so. The PSTN has traditionally used ulaw/alaw and fax
machines are designed to work fine over it.
-Rusty
___
--Bandwidth and Col
On 3/28/06, Rusty Dekema <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I am trying to get Asterisk 1.2.6 to run on Solaris 8 (sparc). I was
> able to get it to compile, but when I try to start asterisk
> (./asterisk -cvg from /opt/asterisk/usr/sbin), I get the following
More info:
Hi,
I am trying to get Asterisk 1.2.6 to run on Solaris 8 (sparc). I was
able to get it to compile, but when I try to start asterisk
(./asterisk -cvg from /opt/asterisk/usr/sbin), I get the following
error:
(snip)
Asterisk Dynamic Loader Starting:
[res_musiconhold.so]Mar 28 09:23:48 WARNING[1829
On 3/15/06, Scott Plante <[EMAIL PROTECTED]> wrote:
> Hi,
>
> We're using Asterisk to develop a specialized IVR system for our
> employees and someone is telling us there is some OSHA requirement that
> you have to always be able to reach a "live human" on such systems. I've
> never heard of that a
Hello. I am following the directions in your legal disclaimer because
I received a copy of your message, yet the message was not addressed
to me. My e-mail address is [EMAIL PROTECTED], but the message I
received was addressed to [EMAIL PROTECTED]
I would hate to see your confidential electronic m
On 3/9/06, Adam Robins <[EMAIL PROTECTED]> wrote:
> Can someone tell me what I'm doing wrong here? I'm trying this from the
> command prompt.
>
> # echo "Hello World" | /usr/bin/text2wave -scale 1.5 -F 000 -o
> /tmp/1141915933.wav
> rateconv: failed to convert from 16000 to 0
> doing v
> #
I thi
On 2/28/06, < Arnaud > <[EMAIL PROTECTED]> wrote:
> What are the options to hook a T1 card up to a laptop running * ? Are
> there USB or PCMCIA T1 cards ?
>
> Has anyone tried a USB to PCI adapter as such :
> http://www.mobl.com/expansion/products/cardbus_expansion/1slot/
> It looks nice but cost 1
In the US, Sprint's CDMA network will do the fancy GPS+AFLT business,
but like someone else mentioned, it only sends the location data back
to Sprint's network. There is an API that you can use to access this
data for your handsets, but you have to pay some amount of money for
each location fix.
S
On 2/22/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
>
> On Feb 22, 2006, at 10:24 AM, Rusty Dekema wrote:
> > This sounds like yet another reason to avoid purchasing Sipura
> > equipment and supporting Sipura in any way. I don't know about you
> > guys, but I
On 2/22/06, Matt <[EMAIL PROTECTED]> wrote:
> Yes.. there are provisioning tools that you have to get.
> Unfortunately it's this catch 22 loop. You have to prove that you can
> offer 200+ ATAs to customers, or you can't get the tools, but yet, you
> don't really want to offer those ATAs to the cus
I don't think it takes a great leap of the imagination to infer that
Mr. Kennedy is in fact having the problem he describes and that,
although it may not be 100% standard and correct usage, the question
mark at the end of his sentence is intended to ask why this problem
might be happening.
If you
On 2/7/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> Tower Server with Digium TDM04B (4FXO Card) - Roughly $1000
>
> 8 Port FXS gateway - $600-$1000
(snip)
For an application like this, what would be the advantage of spending
$600-$1000 on an 8 port FXS gateway rather than spending $280 on four
2-
I have found that it takes a certain amount of time for the codec
preferences that I enter on Teliax's website to actually take effect.
The amount of time has varied from 5 minutes to several hours in my
experience, although I have not changed the settings in some time.
I have also found that if T
I, for one, am glad this message was posted because I was about to call Digium and try to RMA my TDM400B card. Calls using it (in and out) have stopped working as of today (although pure-VoIP calls seem to work fine) for absolutely no reason that I can ascertain. I am about to upgrade to
1.2.3 and
Is anyone else experiencing trouble with Teliax? I can only intermittently register to, and am not able to place any outgoing calls through my assigned gateway; voip-co3.teliax.com.
-Rusty
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On 1/10/06, Carlos Alperin <[EMAIL PROTECTED]> wrote:
And, as I said before, I'm not a religious man, but I don't like otherpeople trying to be funny with somebody else traditions or believes.Personally, I like being funny about traditions and beliefs a whole lot better than I like being overly ser
Oh, wow, ok. Doesn't look like the problem is with your WAN then! (Assuming that the ping times stay like that when the network is at its normal load.) -RustyOn 1/10/06,
Geoff Manning <[EMAIL PROTECTED]> wrote:
Rusty Dekema wrote:> How far (physically) is the Asterisk server loc
There are many ways in which you could do this, but here is one of the simpler and cheaper ways to do it: You and your friend would each buy what is known as an "FXS to SIP Gateway," such as the Grandstream Handytone 286, which is available for $45. (You may wish to use a slightly more common and
On 1/9/06, Terry H. Gilsenan <[EMAIL PROTECTED]> wrote:
I have phones in the US conencted to and Asterisk box in .AU, the pingtime averages 230ms, (ADSL at both ends) and the call quality is just fine.The problem will be if packets are being dropped, of if one or other of
the end-points is getting
How far (physically) is the Asterisk server location from the location of the phones? Have you tried pinging the Asterisk server from the network to which the phones are connected? As a rule of thumb, If the two sites are within 2500 miles of each other and the network connection between them is wo
On 1/8/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
I'm not sure if the fact that my boss is an open source advocate is a good thing or not... ie yes it's great to work with Asterisk and see all the features coming together (especially with Polycom phones). On the other hand I wonder how useful
$300 seems pretty expensive for such a device, especially since someone using it in conjunction with Asterisk would most likely not need its built-in routing features. It's a nice looking device though! Thanks,
RustyOn 1/5/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
SICPE has a new product called t
No; the disclaimer is still completely senseless. Messages you send to the list will be copied, retained and disclosed all over the web by whoever feels like doing so. Furthermore, you do not address your messages to any of us persons, you address them to
asterisk-users@lists.digium.com.If I were
I believe Trixter meant that laws vary from one place to another, and we have no idea where you are located, where you will be placing the calls from, or where the calls will be placed to. Without that information, nobody will be able to help you, as there is not a single global law governing the t
Sorry, meant the "same or better" cpu than the Treo 650.-RustyOn 12/14/05, Rusty Dekema <[EMAIL PROTECTED]
> wrote:I know it was somewhat tangential to your original point, but if a 416MHz device that probably has the same or lesser Intel XScale processor as the Treo 650 can
I know it was somewhat tangential to your original point, but if a 416MHz device that probably has the same or lesser Intel XScale processor as the Treo 650 can't even keep up with running a sip client, it may be "a challenge" (to put it diplomatically) to get Asterisk running well on that platform
The PPC-6700 (http://www.mobiletechreview.com/Sprint-PPC-6700.htm) that I am testing right now may not be able to run Asterisk, but it sure can connect via
802.11b to my Asterisk system. Unfortunately, the operating system does not seem to provide access to the "earphone" speaker that the regular
If it's an "electronic" device, which this certainly is, and if it works on 100-240V, it will almost certainly work at either 50 or 60Hz. It probably gets converted to DC anyway but even if not, there wouldn't be much point in manufacturing a 100-240V power supply if it wouldn't work on both 50 and
Huh?
I think the original poster might be talking about the opposite of what
you are talking about. I understood him to mean that he wants a device
that stays connected to an Asterisk extension and plays any audio
received over the connection through a loudspeaker. I don't see how a
cassette play
Many companies can do that including the following:
www.teliax.com
www.gafachi.com
www.broadvoice.com
www.sellvoip.net
Of these I would recommend Teliax, although I have not had a lot of experience with any of them.
-Rusty
On 12/7/05, Alvaro Parres <[EMAIL PROTECTED]> wrote:
Hi any one can reco
With regards to #3, the advantage of putting in a PRI is that you get
23 or 24 channels that are absolutely reliable and have signal/voice
quality that is essentially as good as it gets. A PRI has got to be one
of the most reliably engineered basic services that a telephone company
can offer. With
now curious as to whether there is such a
syntax.
Thanks,
RustyOn 11/28/05, Rusty Dekema <[EMAIL PROTECTED]> wrote:
Hello,
I am trying to set up my dialplan in such a manner that calls to
numbers in the form 1-NPA-NXX- will only go through if the NPA
dialed is a geographical NPA
Hello,
I am trying to set up my dialplan in such a manner that calls to
numbers in the form 1-NPA-NXX- will only go through if the NPA
dialed is a geographical NPA in the continental United States.
I have collected a list of all NPAs that I want to allow, and have made
the following dialplan
Hi,
I have noticed that most mobile phones (GSM and CDMA at least) seem to
have a tendency to interrupt the incoming audio stream when the
microphone levels get louder than a certain threshold (such as when you
are speaking into it). I do not know exactly why this happens, nor
whether it is someth
I use G.711u with VoIPJet via IAX2 with trunking enabled. From watching
packet flows with netstat, it appears that trunking is enabled in the
data that my Asterisk machine sends to VoIPJet but not enabled in the
data that VoIPJet sends to my Asterisk machine.
That is to say that if there are two c
Thanks for all the replies. I probably shouldn't have used the word
"string" in callerid string, as I really meant just the numeric digits,
not the calleridname data, as I am aware that there are many
complications behind that :).
-Rusty
___
--Bandwidth
Hey,
Has anybody been able to get Broadvoice to pass the callerid string
that Asterisk feeds it to the PSTN? If not, can anyone recommend a
provider with a similar pricing structure (monthly fee for more-or-less
unlimited termination to USA and 20-30 other countries) that will pass
callerid (prefe
exten => 2000,1,Answer()
exten => 2000,2,MP3Player(filename)
exten => 2000,3,Hangup()
-RustyOn 11/18/05, Amith <[EMAIL PROTECTED]> wrote:
hi all,i'm trying to Stream mp3's when dialing a particularextension.2000 in this case.My last part of extensions.conf is as below :exten => 2000,1,Answer
exten
(Sorry; meant to reply privately.)
-RustyOn 11/17/05, Rusty Dekema <[EMAIL PROTECTED]> wrote:
Oh, you do? How much did you have in mind for that? What band(s) does
it run on? Alternatively, if you have any 800 (850) or 1900 MHz GSM
gateways, I'd be interested in hearing about it t
DMA gateway. Do contact us
for more info if you are interested.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Rusty Dekema
Sent: 17 November 2005 14:45
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM
Gateway / Termin
I wish I lived in a 900/1800 country; I'd be all over this!
Unfortunately I am not ready to move to Europe over a £60 GSM gateway, but thanks anyway.
-RustyOn 11/17/05, Sam Tam <[EMAIL PROTECTED]> wrote:
Single port GSM Gateway support 900 / 1800 GSM mode with external antenna.Brand new unit and
I can't build the libiax2 that comes with it on my FreeBSD system :(. Unfortunately FreeBSD is not a supported platform.
-RustyOn 11/16/05, Lee Howard <[EMAIL PROTECTED]> wrote:
Jonathan k. Creasy wrote:>I am using the libiax2 that I just got out of CVS with "cvs checkout>libiax2">As the README st
Problem:
I bought a TDM400 card bundle with two FXS (green) modules so that I
could connect two analog telephones to my Asterisk-1.2.0-rc2 server.
One of these phones is a relatively new rotary-dial pulse phone in good
working order. The other is a DTMF phone that can also be put into
pulse mode.
No. IAX and SIP are two completely different protocols for sending
voice across IP networks. IAX-Trunking is a feature of IAX, and the SIP
protocol does not have any such method for conserving bandwidth by
combining data from multiple calls into one packet.
-Rusty
On 11/12/05, chawki hammoud <[EMAI
At least the soft limit is explicitly published ("X Minutes") as
opposed to most companies' policy of "There is a soft limit, and we
will not tell you what it is, but if you reach or exceed it we will
[charge you $100/day | terminate your service | switch you to a more
expensive plan without notice
Hi,
Does anyone know if it is possible (and if so, how) to disable the
comfort noise generation "feature" on Grandstream telephones? I
received my first Grandstream phone in the mail today and have been
experimenting with it, and I have found that it works very well but the
comfort noise that it g
Speaking from my own experience, I would say that "the point" is that
when a user has, for instance, a DSL line running at 3.0mbps down and
256kbps up, it is very easy to saturate the outgoing bandwidth,
resulting in queues of up to 1-2 seconds for outgoing packets at the
DSL modem.
The ADSL line
How do you get your system to use IO-APIC style interrupts? I am
running linux-2.6.14 and have enabled "Local APIC support on
uniprocessors" and "IO-APIC support on uniprocessors" in the kernel
options, but /proc/interrupts says that everything is using XT-PIC. I
am running an Intel P3-1200 CPU alt
It's possible that your SPA-2000 is set up to read a configuration file
from a remote host every time it boots up, which would overwrite any
changes you make. If you log in as admin and go to the advanced view,
there is an option under the Provisioning tab called Provision Enable.
Make sure that th
I do it this way:
exten => *, 1, Authenticate(PASSWORD)
exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
exten => *, 3, Hangup
It seems to work fine...
-Rusty
On 11/7/05, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
I'm trying to set-up a dialplan for incoming calls that allows a breakoutby
I should add that I am using ulaw (g.711u) for all calls.
-RustyOn 11/3/05, Rusty Dekema <[EMAIL PROTECTED]> wrote:
Does anyone know if it is possible to configure Asterisk in such a way
that it will ignore RFC2833 DTMF signals received from a SIP peer?
I am using Broadvoice for some D
Does anyone know if it is possible to configure Asterisk in such a way
that it will ignore RFC2833 DTMF signals received from a SIP peer?
I am using Broadvoice for some DIDs at the moment and their system has
a tendency to mis-interpret DTMF digits, especially ones dialed from
mobile or office (d
Hi,
I am planning to connect my Asterisk PBX to one or two POTS lines, and
am wondering if it is better to use a TDM card for this, or one or two
SIP devices with FXO ports on them (such as an SPA-3000, Grandstream
488). I am interested in voice quality and reliability of operation and
am wonderin
Hello,
I am wondering if it is possible to get Asterisk to distinguish between
the situation where you place a call to a PSTN line via a SIP telephony
provider and nobody answers, and where you place the same call but the
line is busy.
Watching the Asterisk console on verbosity=5 reveals that As
Hi,
I apologize in advance if this is a stupid question, but I have not been able to find an answer by searching the web.
I would like to add an FXO port or two to my Asterisk setup, and I am
wondering if there is any good reason to spend $120 on a TDM01B or $180
on a TDM02B instead of paying $9.
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