On 22 Sep 2004, Sudhir Kumar wrote:
> Is there a soft phone for PocketPC or iPaq? If not, is someone working
> on it? I will be more than willing to contribute my mite if needed.
Xten has a product, possibly still in beta.
Peter
___
Asterisk-Users m
Title: RE: RE: Creating conference calls from within Astman.
Sorry about my earlier e-mail. (:blush:), should have included the history.
Can this be done from within Gastman?
Warm Regards
Shad
--
Message: 2
Date: Wed, 22 Sep 2004 21:10:10 -0300
From: Nicol
On Thu, Sep 23, 2004 at 01:10:44AM +0200, Gunther Stammwitz wrote:
> Hello,
>
> I just got my isdn-card working together with i4l and asterisk.
> Everything seems to be working fine: I can accept calls coming from the
> outside and I can dial out. Even setting the msn works like charm but my
> pro
Lenny Tropiano / asterisk.org Mailing list wrote:
These taken tonight (9/22/2004) at the Expo and Reception
Enjoy. http://photos.tropiano.org/gallery/astricon-2004
Lenny
Anyone knows if those Snom Keypad 220s are available, and where I might
be able to get my hands on a few?
Flynn
__
Try SJPhone...
- Original Message -
From: "Sudhir Kumar" <[EMAIL PROTECTED]>
Subject: [Asterisk-Dev] Softphone for PocketPC or iPaq
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To UNSU
The VT1000 runs Vxworks.. There is a header on the board that will
allow you to serial console into the operating system and reconfigure
the unit. You will have to connect a MAX232 chip to it because the
header is TTL and you need to convert it to serial for your computer.
it's 38400Bps
- Jason
Email me and we will mirror you.
Brandon
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On 23 Sep 2004 at 5:13, Kevin Walsh wrote:
> Lenny Tropiano / asterisk.org Mailing list [EMAIL PROTECTED] wrote:
> > These taken tonight (9/22/2004) at the Expo and Reception > Enjoy.
> http://photos.tropiano.org/gallery/astricon-2004 > Could you make that
> website a little bit slower? :-)
>
Th
--- Shaun Ewing <[EMAIL PROTECTED]> wrote:
> On Wed, 22 Sep 2004 10:20:58 -0400 (EDT), Jon Miron
> <[EMAIL PROTECTED]> wrote:
> > Hey all,
> >
> > Wondering if this is possible.. Incoming call is
> > answered through X100P, then an extension is
> dialed
> > using the same X100P card. Basically
Lenny Tropiano / asterisk.org Mailing list [EMAIL PROTECTED] wrote:
> These taken tonight (9/22/2004) at the Expo and Reception
> Enjoy. http://photos.tropiano.org/gallery/astricon-2004
>
Could you make that website a little bit slower? :-)
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
Title: Re: RE: Creating conference calls from within Astman.
Thanks for this information.
Can this be done from within Gastman as well?
Warm Regards
Shad
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I am having the same issues with the inbound calls and Galaxy Voice. I
reported this issue a few weeks ago as well and have been unable to
resolve.
Kevin
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 9:39 PM
To: Asterisk Users Mai
We've got some replies to questions online about Astricon and we now
have a mirror available at:
http://astricon.voctel.com/news.php
If anyone has any comments about Astricon, please forward them to me
and I will put them up on the site so that all the people who didn't
go can read them.
Chee
interesting indeed
- Original Message -
From: "Nathan C. Smith" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Wednesday, September 22, 2004 10:52 PM
Subject: RE: [Asterisk-Users] Can someone suggest
> Maybe you could includ
SJPhone from SJLabs.
www.sjlabs.com
Also, a lot of simple questions like this can be answered by looking at
www.voip-info.org
There is a large section there on different soft/hardware phones.
Regards,
Jamie Carl
Chief 'Stuff' Officer
J-Code International
Email: [EMAIL PROTECTED]
PH: +6141436546
Is there a soft phone for PocketPC or iPaq? If not, is someone working
on it? I will be more than willing to contribute my mite if needed.
Thanks,
-- sudhir
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Maybe you could include something on small-world theory. IF everyone
connected to only 6 other switches we could, in theory, connect the world.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 22, 2004 9:49 PM
To: Asterisk Users Mailing List -
Netweb group spams all the lists
- Original Message -
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: "'SeshKanuri'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List -
Non-Commercial Discussion'" <[EMAIL PROTECTED]>
Sent: Wednesday, September 22, 2004 3:05 PM
Subject: RE: [Asterisk-Users
a world of asterisk switches all interconnected via iax.
- Original Message -
From: "m. smadi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 9:22 AM
Subject: [Asterisk-Users] Can someone suggest
> I am taking a course in nework resource management with a
These taken tonight (9/22/2004) at the Expo and Reception
Enjoy. http://photos.tropiano.org/gallery/astricon-2004
Lenny
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What you said was true. I was trying to keep it simple. We are going to be
doing all our own support and are not dealing with the dist companies you
mentioned.
Right now we are happy to be able to offer the low cost product.
Brandon
> This is part of the Vonage deal. Vonage gets to deal with th
Hello,
I have the following configuration :
E100P -> * -> TDM400 -> Modem
When I receive FAXes, about 20% of them are corrupted : pages are not always
complete. If the fax is complex or with numerous pages, it's usually a mess.
Before that, I was using spandsp with success. Unfortunately it's to
On Wed, 22 Sep 2004, Andreas Anderson wrote:
> i've successfully installed the current bristuff, everything works fine,
> exept one thing:
>
> As soon as i pick up the phone that is connected to the bri-card, asterisk
> jumps into
> extension s of the context that is specified in zapata.conf, i
OK, more info.
All outbound seems to be fine with no problems.
Inbound worked once out of about 10 test calls. The rest got a fast
busy.
I programmed a Grandstream 101 and it didn't work against the new proxy
but did against the old one (perhaps a clue as to the IP change?).
I can get inbound c
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of el Flynn
> Sent: Wednesday, September 22, 2004 7:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Transfering incoming calls using same
Brandon Patterson (peering) wrote:
>Ok guys here is the deal. Someone decided that the boxes were shipped to
>soon. Alright, whatever. Now we are told that Linksys will have a program
>where you must be certifed as an ISP or VOIP company in order to purchase
>the box. Call as many dist. as you lik
On 23 Sep 2004 at 2:30, Patrick wrote:
> On Wed, 2004-09-22 at 20:06, Steve Kann wrote:
> > Try app_conference. In this configuration, you should be able to
> > handle 200++ users without problems. It's ideal for this kind of
> > thing.
> >
> > (it's located in iaxclient CVS at iaxclient.sf.net
On Wed, 2004-09-22 at 20:06, Steve Kann wrote:
> Try app_conference. In this configuration, you should be able to
> handle 200++ users without problems. It's ideal for this kind of
> thing.
>
> (it's located in iaxclient CVS at iaxclient.sf.net).
>
> -SteveK
>
Hi Steve,
Thanks for the tip.
On Wed, 22 Sep 2004 10:20:58 -0400 (EDT), Jon Miron <[EMAIL PROTECTED]> wrote:
> Hey all,
>
> Wondering if this is possible.. Incoming call is
> answered through X100P, then an extension is dialed
> using the same X100P card. Basically I want to dial
> in, enter 9 + and have it do a flash then
That is an incredible difference. I'm going to have a look at it.
Darren Wiebe
[EMAIL PROTECTED]
Steve Kann wrote:
Try app_conference. In this configuration, you should be able to
handle 200++ users without problems. It's ideal for this kind of thing.
(it's located in iaxclient CVS at iaxclien
Hello,
>-Original Message-
>From: Shad Mortazavi
>Sent: Friday, September 17, 2004 1:03 PM
>To: [EMAIL PROTECTED]
>Subject: Creating conference calls from within Astman.
>
>Dear All,
>
>I have a requirement to 'originate' a number of calls to various
external users from within >a con
Dee Lowndes wrote:
Hi All,
I am testing out Asterisk with IAX between 2 machines on local
IP addresses and I want one machine to act as an IAX gateway with the
other connecting to it. Anyone know of or can supply me an example of
how to do this?
Cheers,
Dee
You should look up the Wiki at
h
Jon Miron wrote:
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is dialed
using the same X100P card. Basically I want to dial
in, enter 9 + and have it do a flash then
have it dial *08 + # on the
same PSTN line to have it transfer my call to
Title: RE: Creating conference calls from within Astman.
Dear All,
I sent this question a while back and was wondering if this was possible?
Thanks
Shad
-Original Message-
From: Shad Mortazavi
Sent: Friday, September 17, 2004 1:03 PM
To: [EMAIL PROTECTED]
Subject: Creating con
here works fine.
freebsd 5.2.1-release
On Wed, 22 Sep 2004 19:03:54 -0400, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> hi, do any of you guys using the port from freebsd have other problems?
> the whole thing doesnt work for me, as in, after the first phone calls,
> all calls dont have outgo
I am a newbie to Asterisk (though not to SIP) . I am trying to setup a
pure SIP environment for some testing. Here is my SIP.CONF file:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
[247417]
type=friend
host=dynamic
dtmfmode=inband
secret=xyz123
context=default
And my EXTENSION.CONF
Can you let me know what messages were omitted?
Thanks
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: September 22, 2004 11:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] American vs English
Folks,
A few people h
hi, do any of you guys using the port from freebsd have other problems?
the whole thing doesnt work for me, as in, after the first phone calls,
all calls dont have outgoing audio, also if i have a register line
in sip.conf, outgoing audio wont work at all. it seems that * on
freebsd is really
Hello,
I just got my isdn-card working together with i4l and asterisk.
Everything seems to be working fine: I can accept calls coming from the
outside and I can dial out. Even setting the msn works like charm but my
problem is that I cannot hear a word. There's complete silence in both
directions.
that's my music :)
On Thu, 23 Sep 2004 10:33:30 +1200, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> And the astricon conference call currently consists of bing bong,
> deet doot and the odd bit of inspired ambient music (or is that line
> noise?).
>
> Send us info! Pulllze!
>
> Cheers
Folks,
A few people have made me aware of some omissions in my files (not my
fault, they weren't in the Script from the Wiki) which I shall be
tackling this weekend.
Whilst I'm making the files are there any other files you want? IVR's
etc. If so make sure I have a script sent by email.
--
M
That is really weird... especially considering some of the PAP2-NA's
have already shipped. Mine is already on the FedEx truck due to arrive
tomorrow. I wonder what kind of support they will give customers who
managed to snag one during this little window of opportunity?
-jeremy
Brandon Patte
On Thu, 23 Sep 2004 00:00:01 +0200 (CEST), [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Hello!
>
> Can anyone recommend a good/handy/nice sip soft phone?
SJphone (SIP) http://www.sjlabs.com/
iaxcomm (IAX) http://iaxclient.sourceforge.net/iaxcomm/
Those are the ones I like.
Marconi.
__
And the astricon conference call currently consists of bing bong,
deet doot and the odd bit of inspired ambient music (or is that line
noise?).
Send us info! Pulllze!
Cheers,
Matt Riddell
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php
[EMAIL PROTECTED] wrote:
> Hello!
>
> Can anyone recommend a good/handy/nice sip soft phone?
I have been using and like xlite. It seems to work wthout any major
problems.
> I have already done some testing with kphone and gnome meeting (which
> cant do sip).Can you recommend a open source project
Im new to asterisk and telephony in general...so any help would be
greatly appreciated
I have asterisk system running on a T1 PRI using a T100P.
As far as I can tell all the .conf files are correct but every 10
seconds I get the following:
Begin Debug---
T203 count
steve szmidt [EMAIL PROTECTED] wrote:
> On Sunday 19 September 2004 10:28 pm, C Wegrzyn wrote:
> > I ran the LiveCD version of Asterisk on my hardware and it worked. I am
> > trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps
> > getting a seg fault using the sample configurat
"Shawn Kelley" <[EMAIL PROTECTED]> writes:
>> I am preparing to setup a system using Cisco 7940 and 7960's I have
>> the 7.1 SIP firmware on them. One issue I have run into is how to
>> silence the ringer if a call comes in and you don't want to take it.
>> Many phones have a DND button. I know
Im new to asterisk and telephony in general...so any help would be
greatly appreciated
I have asterisk system running on a T1 PRI using a T100P.
As far as I can tell all the .conf files are correct but every 10
seconds I get the following:
Begin Debug---
T203 count
Hello!
Can anyone recommend a good/handy/nice sip soft phone?
I have already done some testing with kphone and gnome meeting (which cant
do sip).Can you recommend a open source project?
It should mainly be practial and have a address book.
I found kphone quite unstable, the address book is design
> On Tue, 21 Sep 2004 21:49:33 -0400, William Suffill
> <[EMAIL PROTECTED]> wrote:
> > Good idea Matt. Tad far for you unfortunately and too costly for me
> > at this time but hearing all the latest and greatest news would be
> > supper.
Anyone have any photo's or thought yet? Us poor outsiders h
Does the PAP-NA2 work with the Sipura firmware and tftp provisioning
options?
Stephen
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Merkel
> Sent: Wednesday, September 22, 2004 9:07 AM
> To: [EMAIL PROTECTED]
> Subject: [Ast
Hi Guys,
i've successfully installed the current bristuff, everything works fine,
exept one thing:
As soon as i pick up the phone that is connected to the bri-card, asterisk
jumps into
extension s of the context that is specified in zapata.conf, if i have
immediate=no then
i hear the normal dia
I got a call from GV on Monday evening telling me they wanted me to move
my Asterisk server over to a new IP address (216.229.127.40) by this
saturday. Why the couldn't tell me this in an email is beyond me but
anyways ..
So I done changed the number and so far its all ok but whilst testing I
noti
the dev conf is friday from 9am - 4pm EST as far as i know
Any more info would be cool. I think an outline of the topics are on
astericon's site
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To U
I already called them. They don't have any and don't have a eta on a new
shipment. Someone bought the 47 they had in stock this morning :-(
Gary
same here
On Wed, 22 Sep 2004 16:17:08 -0300, Bartosz Jozwiak <[EMAIL PROTECTED]>
wrote:
I would love to have contact info for Bottom Line Tech also.
an automated task (cron) DIAL an external IVR (ex: ticket reservation)
via a ZAP channel (analog line)
this automated task has to
1 - send DTMF codes in response to the IVR
2 - detect the end of each of the IVR voice messages
3 - detect IVR possible hangup
4 - detect some "bips" generated by the IV
same here
On Wed, 22 Sep 2004 16:17:08 -0300, Bartosz Jozwiak <[EMAIL PROTECTED]> wrote:
> I would love to have contact info for Bottom Line Tech also.
> Then we do not have to go with all the trouble getting to them.
>
>
>
> - Original Message -
> From: "Gary Carr" <[EMAIL PROTECT
Hi,
I'm a little bit in * world and I got also everthing installed with an
FXO card X101P:
ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
cat /proc/zaptel/1
Span 1: WCFXO/0 "Wildcard X101P Board 1"
1 WCFX
On Wed, 22 Sep 2004, Joseph wrote:
> This is normal. I had the same question and the answer was, that this is
> a good thing to keep the channels as they should be.
>
> See this comment.
> http://lists.digium.com/pipermail/asterisk-users/2004-January/033709.html
This helps with the situation whe
On Wed, 22 Sep 2004, Tony Mountifield wrote:
> Does anyone know which physical interrupt line out of the four on the PCI backplane
> the TE405P uses? Or is it somehow configurable by hardware or software?
Depends on both the hardware (electrical traces on the motherboard) and
software (bios conf
Way cool :)
I noticed a couple of differences between Grandstream's GAPSLITE tool and
your tool:
1) GS ignores multiple occurrences of a parameter, only using the last.
For example:
P30=time.nist.gov
P30=clock1.redhat.com
GS's tool only puts "P30=clock1.redhat.com" in the cfg fi
> On 21 Sep 2004 at 15:40, Kristian Kielhofner wrote:
> > Hey,
> >
> > I am here at Astricon and about to go down to registration. Is there
> > any
> > interest in pictures if I take my digital camera? I am sure that
> > someone is already doing this. (Probably someone official). I would
> > t
Ok guys here is the deal. Someone decided that the boxes were shipped to
soon. Alright, whatever. Now we are told that Linksys will have a program
where you must be certifed as an ISP or VOIP company in order to purchase
the box. Call as many dist. as you like but, selling the box after being
told
Since upgrading to
7.2, I've noticed a random problem where I dial a number and hear all the
correct tones in the handset, but the display won't show all the numbers I
dialed. So you sit there waiting for the dialplan to kick the call off
(b/c you heard the proper amount of tones played and
On Sunday 19 September 2004 10:28 pm, C Wegrzyn wrote:
> I ran the LiveCD version of Asterisk on my hardware and it worked. I am
> trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps
> getting a seg fault using the sample configuration files. Does Asterisk
> not work with the 2
On Saturday 18 September 2004 06:21 pm, Lyle Giese wrote:
> Perfectly normal. On analog lines, the caller id is set between the 1st
> and 2nd rings. So Asterisk has to wait for the caller id and depending on
> the speed of the computer that hosts Asterisk, 13 seconds is exactly right.
> A normal
Does anyone know which physical interrupt line out of the four on the PCI backplane
the TE405P uses? Or is it somehow configurable by hardware or software?
I'm trying to diagnose a problem where the card generates no interrupts in one
system, but is fine in another system. These systems are SBC-in
On Wed, 22 Sep 2004 10:13:58 -0500, Bob Klepfer <[EMAIL PROTECTED]> wrote:
>
> 3) It could be the motherboard. We're on the cheap here and used
> available components to make our server...worked fine with two x100's,
> then the boss wanted another line. Once I got the damn thing to accept
> them
Michael Bielicki wrote:
IAX2/[EMAIL PROTECTED]
you can connect gsm/g726/alaw or ilbc
server is in NL
Ok, an italian link to nufone astricon conf room
is up & running.
Connect it to:
IAX2/[EMAIL PROTECTED]/meetme
OR
IAX2/[EMAIL PROTECTED]/meetmeq
The first one is to listen & speak.
The second one
This is the guy that i talked to and he seemed helpfull
David Durel
Bottom Line Telecommunications
http://www.shopblt.com/
[EMAIL PROTECTED]
Voice / FAX: (860) 886-1011
Monday - Thursday, 9:00 - 6:00 Eastern Time
as i said before i searched the site and the PAP2-NA is no longer listed.
May be a
"Shawn Kelley" <[EMAIL PROTECTED]> writes:
> I am preparing to setup a system using Cisco 7940 and 7960's I have the
> 7.1 SIP firmware on them.
> One issue I have run into is how to silence the ringer if a call comes
> in and you don't want to take it.
> Many phones have a DND button. I know the
Can I contact you off-list?
Please provide email address.
Yiannis Costopoulos.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of SeshKanuri
> Sent: 22 September 2004 22:41
> To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Shaun
> Ewing'
>
Well just did a search on bottom line and they do not have the PAP2-NA
listed anymore. They may still have them in stock if you call them though.
Sorry
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Millican
Sent: Wednesday, September 22, 2004 3:1
I would love to have contact info for Bottom Line Tech also.
Then we do not have to go with all the trouble getting to them.
- Original Message -
From: "Gary Carr" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Wednesday, Se
Here is the contact info For Bottom Line Tech
Bottom Line Telecommunications
www.shopblt.com
457 Route 164
Preston, CT 06365-8111
(860) 886-1011 / (561) 791-3308
John
-O
SeshKanuri wrote:
We want to beat grandstream both at features and price.
I can sell these industry standard PA1688 Chip enabled phones with IAX2, yes
I said IAX2 (along with SIP, H323 and MGCP and a few more such protocols
already enabled) at bulk rates to anyone interested in them.
So does that
There is an asterisk-biz list for this type of post.
Asterisk-users is the non-commercial forum.
Thanks!
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:
You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want
to order some of these.
Gary
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units
and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no
Folks!
Our Phones are cheap and they are selling well. We have no complaints so
far. These phones are made by ATCOM, 2nd largest maker of VOIP gear in
China. We are ATCOM's US distributors.
We want to beat grandstream both at features and price.
I can sell these industry standard PA1688 Chip en
Stay away from the 7910 if your going SIP. It will not support it.
Lethol
On Thu, 23 Sep 2004 01:45:23 +1000, Shaun Ewing <[EMAIL PROTECTED]> wrote:
> The 7910 does not support SIP. It is SCCP only.
>
> -Shaun
>
>
>
>
> - Original Message -
> From: Henry Devito <[EMAIL PROTECTED]>
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
steve how stable is that ?
On Wed, 22 Sep 2004 14:06:29 -0400, Steve Kann <[EMAIL PROTECTED]> wrote:
> Try app_conference. In this configuration, you should be able to
> handle 200++ users without problems. It's ideal for this kind of
> thing.
>
> (it's located in iaxclient CVS at iaxclient.sf
--On Wednesday, September 22, 2004 14:06 -0400 Steve Kann
<[EMAIL PROTECTED]> wrote:
Try app_conference. In this configuration, you should be able to handle
200++ users without problems. It's ideal for this kind of thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
Is there a link/WIK
We are supposed to have some of these on the way. Barring some unforeseen
cicumstance we should have them early next week. E-Mail me off-line if you
want me to set one aside for you.
Michael Crown
Managing Partner
The VoIP Connection
-Original Message-
From: Gary Carr [mailto:[EMAIL PRO
Try app_conference. In this configuration, you should be able to
handle 200++ users without problems. It's ideal for this kind of
thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
-SteveK
On Sep 21, 2004, at 11:13 PM, Darren Wiebe wrote:
I'm presently using meetme extensively on my
Anyone confirmed a stocking vendor we can purchase these from?
Gary
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
T
Hi All,
I am testing out Asterisk with IAX between 2 machines on local
IP addresses and I want one machine to act as an IAX gateway with the
other connecting to it. Anyone know of or can supply me an example of
how to do this?
Cheers,
Dee
___
A
IAX2/[EMAIL PROTECTED]
you can connect gsm/g726/alaw or ilbc
server is in NL
On Wed, 22 Sep 2004 19:56:28 +0200, Brancaleoni Matteo
<[EMAIL PROTECTED]> wrote:
> Ok, an italian link to nufone astricon conf room
> is up & running.
>
> Connect it to:
>
> IAX2/[EMAIL PROTECTED]/meetme
>
> OR
>
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
Thats the first thing I'm going to try when we get our units. I'll g
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
> Hello all,
Sorry for my first mail which answers the 2nd part:(
>
> I'm trying to setup a AVM C2 card.
>
> I have read the kernel requirements for this card.
>
>
> CAPI2.0 support
> [*] Verbose reason code reporting (Kerne
Ok, an italian link to nufone astricon conf room
is up & running.
Connect it to:
IAX2/[EMAIL PROTECTED]/meetme
OR
IAX2/[EMAIL PROTECTED]/meetmeq
The first one is to listen & speak.
The second one is to listen only, use that
if you wanna listen, perhaps with a speakerphone,
in order to not send
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
> Hello all,
>
> I'm trying to setup a AVM C2 card.
>
> I have read the kernel requirements for this card.
>
>
> CAPI2.0 support
> [*] Verbose reason code reporting (Kernel size +=7K)
> [*] CAPI2.0 Middleware support (EXPERIM
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
Ryan Wilkins
On Wed, 22 Sep 2004, Brandon Patterson (peering) wrote:
> This is about big b
Hello everyone. I am trying to do a cvs update. I do the make update; make
upgrade and this is the error that I am getting.
make[1]: Entering directory `/usr/src/asterisk-cvs/asterisk/channels'
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I.
Brandon Patterson (peering) wrote:
This is about big business. No ILEC is going to just sit idle and watch
billions in revenue go out the window. It will be interesting to see if port
blocking ever becomes an issue. Did I buy my Internet service with out
without restrictions? H. Cisco sells to
This really chaps my hide. The situation as it's been explained to me
is: Apparently, too many *consumers* were accidentally buying the
PAP2-NA (unlocked) version and then complaining/returning them to
Linksys b/c they didn't understand that they need a service provider to
be able to place and re
Hello all,
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for
this card.
CAPI2.0 support
[*] Verbose reason code reporting (Kernel size +=7K)
[*] CAPI2.0 Middleware support (EXPERIMENTAL)
CAPI2.0 /dev/capi support
[*] CAPI2.0 filesystem support
CAPI2.0 capidrv i
>
>
> This really chaps my hide. The situation as it's been explained to me
> is: Apparently, too many *consumers* were accidentally buying the
> PAP2-NA (unlocked) version and then complaining/returning them to
> Linksys b/c they didn't understand that they need a service provider to
> be able
This is about big business. No ILEC is going to just sit idle and watch
billions in revenue go out the window. It will be interesting to see if port
blocking ever becomes an issue. Did I buy my Internet service with out
without restrictions? H. Cisco sells to Telco's and Cable guys. Vonage
has
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