On Sat, 14 Aug 2004, Jay Milk wrote:
> QOS comes into play only when you have to route the voice-traffic over a
> WAN connection and it has to compete with data going over the same link.
> If you have a T1 coming into the office and place a VOIP call, then
> someone downloading a huge file needs t
Congratulations, you just volunteered.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Saturday, August 14, 2004 2:04 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] List traffic/Software
>
>
> I know this has been gone over before but
>
> It se
Network traffic really shouldn't be much of a concern if you have decent
n-way switches installed -- pretty much all "mainstream" switches these
days use n-way (Address Table, etc... Lots of names for the same
technology). Those switches basically have enough smarts to know where
the packets are g
On Sat, 14 Aug 2004, Andrew Kohlsmith wrote:
> On Saturday 14 August 2004 18:29, Peter Svensson wrote:
> > Using a BRI will eliminate echos from the pstn connection. Your ip phones
> > should prevent echos from the local phone connections as well. That way
> > you should not cause any noticable ec
For blind transfers, press flash twice, then press #.
For consultative transfers, press flash once, talk to the other party and
tells him to hangup, press flash again, then press #.
- Original Message -
From: "Dennis Cartier" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, Augus
hi man,
if you are trying to upgrade to the latest version, change the permissions of the file, then to the SIP.cnf file add a line that says image version = , copy that line from the Sipdefault.cnf file, .
If the first workaround does not work, try to downgrade to version 2.3 and the do the up
I have 4 7960's that I am trying to get working but 2 of them will not
update to the SIP image on my tftp server like the first ones did.
i keep getting the error on the phone 'Defaulting CM to TFTP server' like it
isn't seeing the *.bin on the server.
are you supposed to have on of those for eac
I hope its okay to post such a long question, but thanks in advance
for reading it
I have been trying to get asterisk to run, I have played with it. and
searched on the wiki and list to no avail
asterisk compiles succesfully ( yes I installed all dependenys )
I am running gentoo 2004.2 with a 2.4
Lots of luck with that one. You may want to wait until I get some specs
on the xml from my vendor. I'll make sure they get to the wiki.
Heck, I may even try writing some code to do the trick for both of us.
In the meantime, if you're pressed for time, you may want to try a SER -
asterisk combo
I'm a noob yes, however I usually read all the docs before asking dumb
question. Perhaps what is needed is better documentation? :-)
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Saturday, August 14, 2004 2:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
> For those that have been selling telecomm for awhile, its fairly well known
> the business purchasing decision is based primarily on "cost" followed by
> "features". Its also fairly well understood that many businesses will
> list a feature or two as "required" to ensure their favorite vendor is
I'm attempting to do a first-time Asterisk install at home, firstly for
use by my self and my family, and secondly as a learning experience.
I've got a new house, and the previous owners removed all but one (1)
phone jack. So I figured I might as well build a PBX.
Functional goals include sta
Hi,
Thanks for your reply. I will try the script...
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Baker
Sent: Saturday, August 14, 2004 5:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP600 shared l
Well, yes it is. Sorry about that. I didn't even think about the Wiki
since what I was looking for was content. I just googled against the
list thinking that was where I saw it. Thanks!
Cheers,
Wiley
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Sat
The wiki is your friend, found it in under 30 seconds.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold
Under also see:
* Sounddogs http://www.sounddogs.com/catsearch.asp?Type=2 Royalty Free
Music
* FreeMusic http://hebb.mit.edu/FreeMusic/ Free Classical
Hello All,
Sorry to rehash a question I am sure has shown several time
but I cannot google up the answer from the lists.
Does anyone know where I can get some royalty free, cost
free music for my music on hold?
I saw someone’s post several weeks ago that said that
this exists at
On Saturday 14 August 2004 18:29, Peter Svensson wrote:
> Using a BRI will eliminate echos from the pstn connection. Your ip phones
> should prevent echos from the local phone connections as well. That way
> you should not cause any noticable echo for the remote party. Being all
> digital has its a
On Sun, 15 Aug 2004 00:29:21 +0200 (CEST), Peter Svensson
<[EMAIL PROTECTED]> wrote:
> On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote:
>
> > I'll most likely use a BRI. Do you think this will help to avoid echo?
>
> Using a BRI will eliminate echos from the pstn connection. Your ip phones
>
On Sun, 15 Aug 2004 00:22:42 +0200 (CEST), Peter Svensson
<[EMAIL PROTECTED]> wrote:
> On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote:
>
> > My concern was if I'd have to teach folks how to dial, but I guess
> > that I can still have the option to assign a number that will give
> > immediate
On Sat, 14 Aug 2004, Wiley E. Siler wrote:
> Greg had a great idea in having you set it up and try it. In fact, that
> is exactly how I did mine. I purchase a cheap clone card for $15 and
> used it to test on one POTS line while I tweaked my configuration files
> and got the system validated. I
On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote:
> I'll most likely use a BRI. Do you think this will help to avoid echo?
Using a BRI will eliminate echos from the pstn connection. Your ip phones
should prevent echos from the local phone connections as well. That way
you should not cause any
Hello Francis,
> I'll most likely use a BRI. Do you think this will help to avoid echo?
I could not say as I have never used a BRI and I am pretty new to this
too. I do know that BRI is supported from watching conversations in
this email list and reading online. People seem to use it a bit so i
On Sat, 14 Aug 2004, Francis Augusto Medeiros wrote:
> My concern was if I'd have to teach folks how to dial, but I guess
> that I can still have the option to assign a number that will give
> immediate access to the PSTN, so no need to make a special dialplan to
> acomodate the weird numbering sy
I read a few discussions on installing Zaptel modules in Fedora Core 2 with
2.6.5 kernel. I was wondering if there is a definitive FAQ on this? I am
still unable to install by FXO card in my pbx box because the modules won't
install.
Thanks
Vikas
_
Hi there Wiley!
On Sat, 14 Aug 2004 14:43:05 -0700, Wiley E. Siler <[EMAIL PROTECTED]> wrote:
> My office build is the same as yours. 15 or so extensions, low traffic
> 100MB network, and a desire to have a phone system that uses VoIP. I
> have my system working as a PBX just like you would. I
Dear Greg,
Thanks a lot for your e-mail! Here are my comments:
On Sat, 14 Aug 2004 14:37:08 -0700, Greg Broiles <[EMAIL PROTECTED]> wrote:
> Asterisk should work fine for this application - but you and/or your
> users may be expecting the Grandstreams to look/act like traditional
> key system pho
No luck on that here either, but the latest sip and firmware updates
have added xml mini browser capability to the IP600 phones. That means
you should be able to add xml code like this:
http://cvs.largeone.net/index.cgi/*checkout*/asterisk/scripts/status.cgi
and get the status of other lines.
J
Use the 'send' button
On Jul 29, 2004, at 3:26 PM, Kanuri, Seshu wrote:
I have setup Grandstream to connect to my Asterisk Server. All the
digits 0-9 are accepting dtmf. But When I try to send the call by
Pressing # Key, nothing happens. Does anyone has a standard
configuration for Asterisk and
Hello Francis,
My office build is the same as yours. 15 or so extensions, low traffic
100MB network, and a desire to have a phone system that uses VoIP. I
have my system working as a PBX just like you would. I use two TDM400s
for my 8 POTS lines and Polycom IP 500 phones at the desktop. I also
Asterisk should work fine for this application - but you and/or your
users may be expecting the Grandstreams to look/act like traditional
key system phones, where you've got a bunch of buttons labeled
"Computer Room" or "Joe" and "Bob", or whatever, where you can press
that button to call that exte
I know this has been gone over before but
It seems that most of the traffic to this list is the same 10 questions
being asked over and over again. What if someone (with some programing
skill) wrote a script so that when someone posted to the list, it would
search the wiki and google and respo
Hi there everyone!
I work at an office where we plant to have about 12-15 phone
extensions. Costs of PBX are cheaper, but they are not expandable and,
as the office is brand new, I want to use all modern stuff.
My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and
install and aster
Please don't post in HTML; use plain text instead.
Ildar Gabdulline <[EMAIL PROTECTED]> wrote:
>
> I'm new to Asterisk and there are several questions on it:
> 1. does it support (or will it support) Fax in the neareast future ?
Yes, but you need to add a component called spandsp.
> 2. will be
Hi,
I'm new to Asterisk and there are several
questions on it:
1. does it support (or will it support) Fax in
the neareast future ?
2. will be Windows version developed
?
3. Are there any known addons that allow to use
Asterisk as:
IP-centrex system
call center
solution
comple
Yes, it helps quite a bit. It shows me where Comedian Mail spawns the
external app.
Do you have a copy of your SIP MWI script? I may be able to use it as a
starting point.
Also, can you tell me what variables are passed from asterisk to the
app?
Thank you very much.
Greg
> -Original Mess
Are you going to use single E1 line? How many concurrent calls.
Single Xeon 3.0 with *1 GB *ram should serve 30-32 calls with not
problem, even during G729 transcoding and echo cancellation.
If you are not plaining to do G729 transcoding - then I believe you can
put more then 32 calls in this box
On Sat, 14 Aug 2004, administrator tootai wrote:
> Hi list,
>
> I have SIP clients and H323 GK connected through h323 channel (Nufone).
> In h323 conf I gave prefix=09 so all call starting with this prefix are
> send to asterisk. The context is also given their as [fromh323]
>
> But now, in asteri
Unfortunately the Cisco ATA-186 does not support iLBC which means
extra costs for purchasing 729 licenses. The ATA-286 works fine other
than this 1 issue.
Do the Grandstream developers follow this list?? This problem has been
persistent for a LONG time and each new firmware version still has it
un
Hi list,
I have SIP clients and H323 GK connected through h323 channel (Nufone).
In h323 conf I gave prefix=09 so all call starting with this prefix are
send to asterisk. The context is also given their as [fromh323]
But now, in asterisk, I want to have the called number without this 2
leading
Yes, we are experiencing the same problem and because of that we
switched the called HT ata to Cisco ATA 186 ...
Lubo
Andy Lee wrote:
[ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ]
The problem I'm experiencing with many GS adapters, regardless of
firmware version is this. Call from one p
[ Message pasted from Sun, 7 Mar 2004 08:27:54 -0600 ]
>
> The problem I'm experiencing with many GS adapters, regardless of
> firmware version is this. Call from one phone to another phone using
> both the 'T' and 't' flags in the Dial() command. After they are
> connected, you should be able to
I have done something simmillar, but not the same.
I send mwi notification to our softswitch (SIP).
Basically I wrote a small app in pascal that sends a
sip message to the softswitch. The app is called
everytime a message is left or retrieved, using the
extrennotify option in voicemail.conf.
Yo
All set up and working now. My problem was that the cable from the E1 on
the patch panel to the TE410p was too long. Also, there was no need to use
DNIS, a standard entry matching the dialed number to the SIP extension was
all that was necessary. Will play with using macros later once everything
On Sat, 14 Aug 2004, Robert Hajime Lanning wrote:
> This is called an Analog DID Trunk. Yes, Asterisk supports it.
> You can have inbound calls only on this type of line. Also, it
> does not support CallerID.
>
> The carrier's CO acts like a POTS analog handset. When a call comes
> in it simul
> Also known as DID service or called number information at various
> times. You can have analog copper pots lines configured to send that
> information. I don't know if Asterisk supports it. Anyone?
This is called an Analog DID Trunk. Yes, Asterisk supports it.
You can have inbound calls only
Hi Ryan!
Interesting what experience you have made in this issue.
We have setup the alternative channel for H.323 (the * built in
chan_h323), and we are now in a testing phase.
I was wondering (in case no transcoding is needed), how your setup treats
the RTP streams. Do the RTP streams go end-to
Hi,
> -Original Message-
> I am setting up an Asterisk system with Cisco 7960 phones. I
> have a PoE injector to insert between the patch panel and HP
> 2626 switch. I plan to plug the users pc into the phone and
> the phone into the wall. I would like the phones to have a
> sepera
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