Yitzhak Bar Geva wrote:
> How can one measure the effect of NAT traversal packet loss?
> We currently have no solution for NAT traversal for our SIP clients.
We've recently completed a setup (see other threads) with a couple of
SIP clients behind NAT in their respective home-offices. Took a coup
Jared Smith wrote:
> On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote:
>> The SPA-9x1 does support http download, but I don't see how you could
>> change the initial TFTP request to HTTP without manually configuring
>> the phone. Even then I'm not sure it would work - I certainly
>> haven't ma
The CFP for HITBSecConf2008 - Dubai is now open.
Our 2008 event is expected to attract over 300 attendees from around the
EMEA region and will see keynote speakers Bruce Schneier (Founder and
CTO, BT Counterpane) and Jeremiah Grossman (Founder and CTO, White Hat
Security). The event is supported a
If you want to do this automatically, what you're looking for is a
(Predictive) Dialler for Asterisk. There are a few available, both on the
commercial and the free side. I'd start by checking out ViciDial /free)
and SineDialer (commercial) that are some of the most used ones.
Thanks
l.
On
The Xorcom Astribanks are quote good - have you looked at those?
PaulH
On Tue, 2007-10-23 at 12:41 +0800, Rilawich Ango wrote:
> What do you mean by interruption? Is it possible to better control to
> prevent it? The options you provided is over my budget. That's why
> I am looking for mult
Rilawich Ango wrote:
> What do you mean by interruption? Is it possible to better control to
> prevent it? The options you provided is over my budget. That's why
> I am looking for multiple TDM cards.
>
> On 10/22/07, Gergo Csibra <[EMAIL PROTECTED]> wrote:
>> Monday, October 22, 2007, 9:47:49
Hey Mike,
We started deploying exclusively Polycom and Linksys. The Polycom's
support presence, they call it 'Buddy List'. I am not sure about the
Linksys phones, I don't think they do although I did see support for
SLA (Shared Line Appearance).
Omar
On 10/23/07, Michael J. Liberatore <[EMAIL PR
What do you mean by interruption? Is it possible to better control to
prevent it? The options you provided is over my budget. That's why
I am looking for multiple TDM cards.
On 10/22/07, Gergo Csibra <[EMAIL PROTECTED]> wrote:
> Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote:
>
> > Hi al
I also have problems with these phones. I have deployed many of them
and have had nothing but problems. Omar, what phones did you switch to?
I needed some of the features of the snom phones, like the multiple
buttons with prescence lights.
Mike
-Original Message-
From: [EMAIL PROTECT
CrossPlatform Linux, Windows, Mac OpenSource WebHuddle at
http://sourceforge.net/projects/webhuddle
It has built in VOIP of some kind, don't remember the details. But why not
use Asterisk or one of the free teleconference websites for the audio and
WebHuddle for the webcams and desktop sharing.
I used to deploy these phones, it was these types of issues that
forced me to drop it. It took way too long to troubleshoot the
problems and there was a general lack of documentation. This was 2
years ago, things might have changed. If I remember correctly, it was
this issue you are having that was
Carlos,
No solo para enviar llamadas, sino tambien para recibir (de hecho, ese
bloque que puse ahi lo uso para recibir, no para enviar).
Te posteo un ejemplo del ejemplo que trae asterisk de sip.conf
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP add
What we found is that even if you get the lights working, they go off
after a few days.
Paul Hales
AsteriskIT
On Mon, 2007-10-22 at 09:49 -0300, Carlos Maimone wrote:
> Dear friends,
>
> I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
>
> In order to get subscriptions working a
There is a way to force the order of the codecs in the sip.conf since the
allow seams to let know only the accepted codec.
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Yitzhak Bar Geva wrote:
> How can one measure the effect of NAT traversal packet loss?
> We currently have no solution for NAT traversal for our SIP clients. There
> is no doubt that packets are getting lost. What is not clear is how much
> damage this
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How can one measure the effect of NAT traversal packet loss?
We currently have no solution for NAT traversal for our SIP clients. There
is no doubt that packets are getting lost. What is not clear is how much
damage this does. On the face of it, everything seems fine. Could this be
so? Perhaps we'r
Anyone managed to get this to work? What's the recipe?
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At 02:18 PM 10/22/2007, you wrote:
>;nothing displayed
>exten => s,n,Verbose(${CALLERID})
>exten => s,n,Verbose(${CALLERIDNAME})
>exten => s,n,Verbose(${CALLERIDNUM})
>exten => s,n,NoOp(${CALLERID})
>exten => s,n,Verbose(${CALLERID})
>
>;CID at last!
>exten => s,n,Verbose(${CALLERID(num)})
>===
Dear Marc;
I readed your email about the codec G729a and I am now
also need to install the codec on my Asterisk.
I typed from Asterisk CLI:
core show version and I got the following:
Asterisk SVN-branch-1.4-r72556 built by root @
localhost.localdomain on a i686 running Linux on
2007-06-30 13:0
On Mon, 22 Oct 2007 17:57:44 -0400, Jared Smith <[EMAIL PROTECTED]>
wrote:
>Beginning with Asterisk 1.4, we moved all of the CallerID functionality
>from channel variables and applications to a single CALLERID dialplan
>function. This should have been noted in UPGRADE.txt. I also tried to
>warn y
Hi,
Iam dialing from NT ptp to SIP provider.
Sometimes Asterisk is doing music on hold but there are no options like t or T
in the dial command. As an result the channel got lost and an Hangup occurs.
Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card.
Any solution for this?
Oct 22 11:20:06
La configuración de Jose esta correcta. Cuando usas un "peer" en
sip.conf Asterisk usa el hostname or el IP para autenticar. Cuando
usas un "user" la autenticación se basa en el usuario y la
contraseña, cual en su caso no existe.
On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> Ho
El Mon, Oct 22 de 2007 a las 15:59 -0500, Carlos Chavez comentaba:
> Hola José. Gracias por tu contestación. Lo que me estas especificando
> el para hacer llamadas de salida (PEER). Yo necesito autentificar a un
> usuario de entrada, voy a intentar haciendo algo parecido solo cambiando
> a
On Mon, 2007-10-22 at 15:13 -0400, [EMAIL PROTECTED] wrote:
> On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> > I have a customer that needs an Asterisk server to sell minutes for
> > cell phones in Mexico. I do not see a problem with that since he will
> > get the calls by SIP and
On Mon, 22 Oct 2007 16:41:19 -0500, "Erik Anderson"
<[EMAIL PROTECTED]> wrote:
>Version 2 of TFOT was just released a few weeks ago...
Just had to ask :-) Thanks.
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On Mon, 2007-10-22 at 23:18 +0200, Vincent wrote:
>
> exten => s,1,NoOp(Got a call)
>
> ;nothing displayed
> exten => s,n,Verbose(${CALLERID})
> exten => s,n,Verbose(${CALLERIDNAME})
> exten => s,n,Verbose(${CALLERIDNUM})
> exten => s,n,NoOp(${CALLERID})
> exten => s,n,Verbose(${CALLERID}
On 10/22/07, Vincent <[EMAIL PROTECTED]> wrote:
>
> 2008 might be a good year to update "* - The future of telephony" :-)
Version 2 of TFOT was just released a few weeks ago...
http://downloads.oreilly.com/books/9780596510480.pdf
--
Erik Anderson
http://andersonfam.org
Gergo Csibra schrieb:
> Well, using more than one TDM card in your PC is not a good idea,
> because of interrupts. If you have to have 16 FXO you can more
> options:
>
> 1. Using TDM2400P with 4 FXO modules ($1775)
> 2. Using Xorcom's Astribank (external) ($1170)
> 3. Using some T1/E1 card with Cha
On Mon, 22 Oct 2007 09:06:00 +0200, randulo <[EMAIL PROTECTED]>
wrote:
>The first ten sites that come up, including voip-info.org, usually a
>good place to look first, each have full examples. Look also for the
>background application wich is used to play the file, get input and
>jump to the exte
On Mon, 22 Oct 2007 21:19:27 +0200, Vincent
<[EMAIL PROTECTED]> wrote:
>Does Zaptel support those on Digium TDM400 clones like those from
>OpenVox?
Pff, finally found what it was: It had nothing to do with zaptel, and
everything to do with extensions.conf:
exten => s,1,NoOp(Got a call)
I have built an asterisk server with a TE412P card on a Dell 2950.
It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions,
Fax/Analog extensions via an old PBX via PRI, voicemail, etc.
My issue now is that I find it difficult to test/upgrade to new versions.
This is what I am think
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard wrote:
> Saludos Carlos,
>
> Como vas a recibir las llamadas via SIP, puedes especificar el IP del
> host que te enviara las llamadas, por ej.
>
> Este es un bloque que tengo definido en el SIP.conf de uno de mis
> servers para enrutar las llamad
Is it possible to implement a dial-out call queue in Asterisk?
My idea is to give Asterisk a list of numbers, and then he makes the
calls and delivers the calls to a call queue.
Then, the agents will answer the calls.
Is this possible?
Thanks
Regards
Joao pereira
___
On Mon, 22 Oct 2007 10:14:41 -0400, Jared Smith <[EMAIL PROTECTED]>
wrote:
>Instead of ${callerid} here (which probably isn't working for you
>anyway), you probably want to use the CALLERID dialplan function to
>retrieve the CallerID number, like this:
Thanks for the tip. It'll come in handy... on
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar las llamadas internacionales y a telefonos moviles
utilizando un proveedor de
Hello
I've been googling for a couple of days now, but still can't
figure out what to put in zapata.conf to get it to report CID.
Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202
as CID FSK Standard:
http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg
http
On 10/22/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> I have a customer that needs an Asterisk server to sell minutes for
> cell phones in Mexico. I do not see a problem with that since he will
> get the calls by SIP and then use GSM adapters to get the calls into the
> GSM network. My
Jon Weisman wrote:
> Here's what worked:
>
> exten=>_X.,1,Dial(SIP/"prefix"[EMAIL PROTECTED] trunk)
>
> substitute "prefix" for the tech prefix you would like to append.
> - Original Message -
> From: "Philipp Kempgen" <[EMAIL PROTECTED]>
> To: "Asterisk Users"
> Sent: Tuesday, Octobe
Hi List,
I am using a Plantronics CS50 head set with my Polycom 601. I use the button on
it to pick up calls. Is there any way to have the phone set up that if I pick
up with the button on the headset that it sends the call to the headset and
that I don't have to press the headset button on the
I have a customer that needs an Asterisk server to sell minutes for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP and then use GSM adapters to get the calls into the
GSM network. My problem is that his customers only want to be
identified by IP
I have built an asterisk server with a TE412P card on a Dell 2950.
It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions,
Fax/Analog extensions via an old PBX via PRI, voicemail, etc.
My issue now is that I find it difficult to test/upgrade to new versions.
This is what I am think
Is this free? I see the tuner is free.. but the speech rec isn’t?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent: Monday, October 22, 2007 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Makin
Check out again http://spc.pifiu.com it seems the owner of the site
has added the latest admin guide for SPA-900 series & the spc.exe for
5.1.5 & 5.1.7 firmware.
On 10/21/07, Per Jessen <[EMAIL PROTECTED]> wrote:
> [EMAIL PROTECTED] wrote:
>
> > If you are trying to use non-complied ("XML") profil
Here's what worked:
exten=>_X.,1,Dial(SIP/"prefix"[EMAIL PROTECTED] trunk)
substitute "prefix" for the tech prefix you would like to append.
-Jon
- Original Message -
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re:
no that didnt work.
- Original Message -
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X. => {
Dial(tech/123{EXTEN});
}
> Luki, thanks for writing to say it DOES work. I've have just now had
> another look, found my mistakes (basically $MAC instead of $MA), and
> it's working!
I'm glad you got it sorted out. Yes, it works with XML or compiled
files. To help with troubleshooting, specify a syslog server and set
the
Nice job! I took the liberty to post it on AstPligg as well:
http://tinyurl.com/268bac
Thanks
l.
In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith <[EMAIL PROTECTED]>
ha scritto:
> On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
>> I’m interested in what software (Free or course) that peo
Hi,
We have done a video and voice conferencing application but it's still
Alpha. We use Red5/Flash for video, IAX for audio.
You can take a look at http://code.google.com/p/blindside/ and click on the
screencast and Webconference demo.
Maybe we can work with each other to further improve it.
R
Coool... thanks man.. do you have any installation procedures or notes?
Thanks!
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Monday, October 22, 2007 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [a
On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
> I’m interested in what software (Free or course) that people use when
> they want to add a “dial by voice” service to their asterisk system.
> Meaning I pick up the phone.. dial some extension… it prompts me for
> name.. I say “John Smith”.. and it
On Sun, 2007-10-21 at 17:22 +0200, Vincent wrote:
> ;here, rewrite CID name by looking up CID # in database
> ;put CID name + number in variables
> ;exten => _[1-4],n,SetVar(cid=${callerid})
> ;send e-mail with CID name + number and link to WAV file to people in
> charge of selected software
Inst
> John Millican wrote:
> > Hello All,
> > I am looking at doing some video conferencing with SIP. I was hoping to
> > get some early pointers from any one that is currently doing this. I
> > have been all over goggle and voip-info and there is a ton of anecdotal
> > information but, I was hopin
[EMAIL PROTECTED] wrote:
> I have an trixbox(asterisk) software on a pc home edition.
> Origination is a Nortel ,model=CS2K,version=ISN08
> and my asterisk is doing termination.Nortel sent calls
> to us ,Asterisk and they said that is sending call
> and i saw the trace as following:
>
> sip: [EMAI
On Sun, 2007-10-21 at 13:42 +0200, Per Jessen wrote:
> The SPA-9x1 does support http download, but I don't see how you could
> change the initial TFTP request to HTTP without manually configuring
> the phone. Even then I'm not sure it would work - I certainly haven't
> managed to make any of my SP
Hi,
I'm interested in what software (Free or course) that people use when they
want to add a "dial by voice" service to their asterisk system. Meaning I
pick up the phone.. dial some extension. it prompts me for name.. I say
"John Smith".. and it dials his extension and connects the call..
Direct single line video conferencing via SIP is actually pretty
straightforward and works rather well.
Multipoint conferencing is where you get into a bit of a mess. There
are precious few products out there that claim multipoint SIP video
conferencing capability, and we've had no luck so fa
Hello All,
I am looking at doing some video conferencing with SIP. I was hoping to get
some early pointers from any one that is currently doing this. I have been
all over goggle and voip-info and there is a ton of anecdotal information
but, I was hoping for more specifics of what people are ac
Lees, James (UK) wrote:
> Hello Again,
>
> I was just wondering if anyone can give me a heads up regarding the
> possibility of identifying that a user currently in an active call is
> also being dialled by another extension.
>
> Does asterisk/sip issue an event that says there's a call attempti
If your SIP phone supports multiple appearances for a line, you should
just get another INVITE coming in while you are on your current call.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Lees, James (UK)
> Sent: 22 October 2007 13:40
> To: ast
Dear friends,
I am working around with a Snom 360 and Asterisk 1.4 + FreePBX
In order to get subscriptions working and the Snom 360 lights turns
on, I have set everything just like all the pages in the net explain.
So, I get subsciption working. I can list subscription on the
asterisk and if
Ciao Andrea,
> I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens
> that I send a request and I receive a response to ANOTHER request that it got
> in the frame time between my request and my response.
>
> Did anyone else notice this behaviour? How can this be solved?
Hello Again,
I was just wondering if anyone can give me a heads up regarding the
possibility of identifying that a user currently in an active call is
also being dialled by another extension.
Does asterisk/sip issue an event that says there's a call attempting to
reach you? If so, I will then u
Hello *,
I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens
that I send a request and I receive a response to ANOTHER request that it got
in the frame time between my request and my response.
Did anyone else notice this behaviour? How can this be solved?
I've been read
I have an trixbox(asterisk) software on a pc home edition.
Origination is a Nortel ,model=CS2K,version=ISN08
and my asterisk is doing termination.Nortel sent calls
to us ,Asterisk and they said that is sending call
and i saw the trace as following:
sip: [EMAIL PROTECTED] IP:5060 ;user phone
but
Monday, October 22, 2007, 9:47:49 AM, Rilawich wrote:
> Hi all,
> I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404
> or 2 TDM808 to get 16 FXO? What is the difference (in performance and
> control) in using 4 x TDM404 and 2 x TDM808 if possible?
> ango
Well, using more than o
Hi all,
I want to have a 16 FXO in a PC. Is it possible to use 4 x TDM404
or 2 TDM808 to get 16 FXO? What is the difference (in performance and
control) in using 4 x TDM404 and 2 x TDM808 if possible?
ango
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On this machine its the first install, but i get this error 3 month before
on an other machine also.
I think the debug will bring t much data, cause there is any half
second a call try, and its really hard to find this error in the debug
file.
The only thing i know is if i use a Sangom
Per Jessen wrote:
> Luki wrote:
>
>> Here's how you do it.
>>
[snip]
>
> Oh well - I wonder what I'm doing wrong then. I've been trying to get
> this to work for most of last week.
Luki, thanks for writing to say it DOES work. I've have just now had
another look, found my mistakes (basically
On 10/20/07, Vincent <[EMAIL PROTECTED]> wrote:
> I've never built an IVR before, so I was wondering if someone
> could share some code from their extensions.conf that would perform
> some of thoses steps:
Try Google for asterisk ivr
The first ten sites that come up, including voip-info.o
We have written stuff previously for most major phones that does
auto-deploymentserver sits there waiting for phone to ask for
configs, when the phones hit the server, the configs are written on the
fly.
Bit fiddly to write, but once it's going it's pretty good.
PaulH
On Sat, 2007-10-20 at
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