On Thu, 21 Oct 2004 23:51:05 +0200, Damjan <[EMAIL PROTECTED]> wrote:
> It seems that the current practise is that until a court validates a
> patent its not really "in force". The reason for this is that patent
> offices have not done their job and grant most of the patent
> applications, so its u
On Thu, 21 Oct 2004 23:55:15 +0200, Damjan <[EMAIL PROTECTED]> wrote:
> If I'm not mistaken, if you patent a device (aparatus) you can't be so
> vague in the patent wording and have to submit a prototype too...
> doesn't that mean that if my device doesn't in any way resemble the
> prototype I'm cl
On Fri, 22 Oct 2004 00:24:30 +0200, Damjan <[EMAIL PROTECTED]> wrote:
> I just think its good to state that it may not be *legally* in force.
No it's not good to say that because it's nonsense.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB:
On Fri, 22 Oct 2004 09:47:21 +0300, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> hmm... now tring.. somone to know how can I redirect the output of the "sip debug"
> into file 'cause it is really hard to grasp (several pages is just one call)
if you are running a fairly recent version of asteris
On Fri, 22 Oct 2004 09:58:55 +0200, Roy Sigurd Karlsbakk
<[EMAIL PROTECTED]> wrote:
> > It seems that the current practise is that until a court validates a
> > patent its not really "in force". The reason for this is that patent
> > offices have not done their job and grant most of the patent
> >
On Fri, 22 Oct 2004 10:46:22 +0200, Philippe Dhont (Sea-ro)
<[EMAIL PROTECTED]> wrote:
> 1) if i setup asterisk, do i need additional hardware if other computers
> via internet want to talk to me ?
if all you want is IP based telephony only, then all you need is
ethernet, LAN wiring, hub, router
On Fri, 22 Oct 2004 10:50:12 +0800, Ronald Wiplinger
<[EMAIL PROTECTED]> wrote:
> I would like to setup Asterisk PBX for following purposes:
>
> 1. dial-in from Europe via sipgate (Germany & UK)
> 2. dial-in from Canada via primus
> 3. dial-in from USA via vonage
> 4. dial-in via two analog lines
On Fri, 22 Oct 2004 14:11:53 +0300, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> thanx, now it works..
Good. Remember, debug is always your best friend. Other feedback may
lie to you (ie "everybody is busy" is most often a lie and it really
means "No clue why it's not working") but debug will al
On Fri, 22 Oct 2004 11:19:56 +0800, Garry Taylor <[EMAIL PROTECTED]> wrote:
> How to set fwd iax as a peer? The config that I got from FWD does not show
> how to do this
Yeah, the sample config of FWD is BAD and should be AVOIDED.
> Also how to get the status from "unmonitored" to what you have?
On Fri, 22 Oct 2004 18:06:27 +0200, Yves-Marie CRABBE <[EMAIL PROTECTED]> wrote:
> I tried to define a disconnect tone description this way :
> [EMAIL PROTECTED],[EMAIL PROTECTED];2(.5/.5/1+2)
> I'm located in France.
Try the localisation wizard on Voxilla.com. France should be on the list.
rgds
On Fri, 22 Oct 2004 18:19:33 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Basically, a patent could be granted in the USA and, by way of
> treaty, be registered in all signatory countries. The patent may
> not actually be valid in all of the countries mentioned in the patent,
> depending upon t
On Fri, 22 Oct 2004 13:10:38 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> How could I test if my ports 1 to 2 are open on my firewall?
nmap is your friend
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages u
On Fri, 22 Oct 2004 20:52:40 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Your views are biased upon the ridiculous legal system used in the US
Well, I have had no exposure to the legal system in the US, since I
never lived there. I did live in the UK though and I did have exposure
to the legal
On Fri, 22 Oct 2004 17:07:11 -0300, Damian Rosales <[EMAIL PROTECTED]> wrote:
> The best explanation I could find whas that a standard modem is half-duplex
> and the connection needs to be full-duplex. Is that all? I wouldn't mind
> having to talk in a half-duplex mode if that's the big issue.
Do
On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson <[EMAIL PROTECTED]> wrote:
> I would appreciate your opinions on the feasibility of these
> techniques, and also about any other methods that have been
> tried to achieve direct SIP connectivity.
If you are that desperate to use Vonage, then why d
On Fri, 22 Oct 2004 16:07:08 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
>
> Do you have a list of those providers that use IAX?
check the Wiki ...
http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers
in addition to those entries which mention IAX there are also some
On Sat, 23 Oct 2004 02:16:44 -0700, Trevor Peirce <[EMAIL PROTECTED]> wrote:
> G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
> G723 - - - - - - - - - - -
>GSM - - 2 2 4 2 1 1238 - - 529
On Sat, 23 Oct 2004 15:06:15 +0200, Yves-Marie CRABBE <[EMAIL PROTECTED]> wrote:
> Thanks a lot, I tried, the string for the busy tone (from Voxzilla) is :
> [EMAIL PROTECTED];10(.5/.5/1)
> I also tried with [EMAIL PROTECTED];4(.5/.5/1) for a shorter detection delay (my pstn
> provider
> doesn't pl
On Sat, 23 Oct 2004 17:10:40 +0200, Neal Nelson <[EMAIL PROTECTED]> wrote:
> I'm using IAXComm on the Mac to connect to my Asterisk system and it
> all seems to work well when I'm connected to my wired network. When I
> use wireless instead, IAXComm never registers with Asterisk and when I
> call,
On Sun, 24 Oct 2004 01:45:19 +0200, Stewart Nelson <[EMAIL PROTECTED]> wrote:
> I looked at NuFone.net and some others, but it appears that
> IAX is not right for my system.
I'd say this is only because you don't know enough about IAX yet ;-)
> I live near Reno, NV, and
> have a second home in Pa
Re: G.729 codec on Yellow Dog Linux for various PPC
Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
> This is probably a good time to ask if there is any
> planned support for a g729 binary for YDL and
> G3/G4, etc.
A customer has ordered some voice prompts from Digium's TheVoice
online store. They say the recordings' sound was good when they
listened to it on their Windoze boxes. However, then Asterisk is
playing back the recordings, the volume is far too high and they sound
really bad. This is particularly n
Hi Steve,
On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> 1. I am sure there are free programs that will allow you to adjust the
> files to sound more like the originial recordings as well as converting them
> to gsm.
that's all very cool, but if you read my post c
On Sun, 24 Oct 2004 14:47:56 +0100, Elliot Moore
<[EMAIL PROTECTED]> wrote:
> Helpful URLS about SOX/wav/gsm
> Have you seen these?
[snip URLs]
yes, I have played with those and all I did achieve was making the
recordings worse, but thanks anyway.
However, it seems now that this is not a common
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco <[EMAIL PROTECTED]> wrote:
> http://www.theivrvoice.com/
>
> would seem to imply otherwise. I'd be a bit surprised if any company had
> enough work to keep her employed full-time, so the "works at Digium" line
> sounds a bit fishy to me.
I thin
On Sun, 24 Oct 2004 15:27:53 +0200, Stewart Nelson <[EMAIL PROTECTED]> wrote:
> No, I don't want to replace existing gear.
fair enough.
> It would be expensive
that I don't agree with, especially not if you do it yourself, but anyway.
> There are other reasons, too. For example, the Cisco 827-
On Sun, 24 Oct 2004 12:01:19 -0300, Nicolás Gudiño <[EMAIL PROTECTED]> wrote:
> I've never ordered from thevoice, but I have converted some MP3
I guess you mean WAV
> to gsm
> and after fighting with sox parameteres I came up with this:
>
> sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -
On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys
<[EMAIL PROTECTED]> wrote:
>
> after a couple of days work banging my head against the wall (bloody standards
> my arse), i've got chan_bluetooth to a point where it's starting to function
> - certianly more than just proof of concept now.
W
On Sun, 24 Oct 2004 21:42:49 +0100, Martin List-Petersen
<[EMAIL PROTECTED]> wrote:
> Regarding the headset, i have not seen how that works yet, but i would
> say you would need to enter the number somewhere (maybe special prefix
> on any phone + phoneno. to get the call to the headset)
Oh well, I
On Mon, 25 Oct 2004 16:55:55 +1300, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> I would say Asterisk would work with any SIP device out there, it would depend
> upon your individual configurations. As long as the device SIP/IAX/H323 it
> should be straight forward to get it plugged into Asteris
On Sun, 24 Oct 2004 18:11:44 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> Has anybody tested any gateways from ACT
Last time those gateways came up in a conversation it was concluded
that efforts should be concentrated on the phones so as to not dilute
engineering resources with too many things. Thi
On Mon, 25 Oct 2004 12:37:33 +0200, Miroslav Nachev <[EMAIL PROTECTED]> wrote:
>Unfortunately the Mediatrix products are very expensive.
just one example. my point was that as of this moment, ACT are more
focussed on their phones and it may well be wise to look for gateways
elsewhere for the t
On Mon, 25 Oct 2004 13:48:35 +0200, Miroslav Nachev <[EMAIL PROTECTED]> wrote:
>Yes, I now all prices of Yoda, and I am looking for some partners
> with which to combine one order for samples.
I am only willing to pay for a sample if they
- GUARANTEE that their product will work 100% in any
On Mon, 25 Oct 2004 11:06:40 -0400, Race Vanderdecken
<[EMAIL PROTECTED]> wrote:
> If you have not gotten your answer yet;
I have tried the sox parameters Nicolas recommended and the files did
seem to be ok with that on my Powerbook. I am now waiting for the
feedback from the customer.
>
On Mon, 25 Oct 2004 20:32:51 +0200, Oscar Bults <[EMAIL PROTECTED]> wrote:
> When I issue the command
> asterisk -c
>
> I can see only a warning for file chan_oss.c: I don't work right with
> non-full duplex sounds cards
try "noload => chan_oss.so" in /etc/asterisk/modules.conf and see if
On Mon, 25 Oct 2004 13:08:37 -0500, David McNett <[EMAIL PROTECTED]> wrote:
> Once you've devised such a mechanism it becomes a simple matter to have
> asterisk route by building an AGI script which runs in the dialplan. The
> simplest approach would probably be to create an intranet web page wher
I am experiencing a rather strange problem.
exten => s,1,Answer
exten => s,2,Background(welcome)
... other auto attendant stuff ...
will result in the calling party not getting any audio (neither
ringing feedback nor IVR audio) if callling in from the PSTN on an
X100P.
However ...
exten => s,1,
Has anybody had any dealings with Yoda Communications in Taiwan?
I am asking because we are at a crossroads with them. We had agreement
that they would ship us a free sample of their 4 port FXO gateway,
that we would carry out testing on the Japanese PSTN and various other
connectivity scenarios c
On Wed, 27 Oct 2004 10:14:29 -0500, Christopher L. Wade
<[EMAIL PROTECTED]> wrote:
> Over the years, we have gotten approximately $100,000 worth of
> support/customer relations calls (counting 800 number fees,
> number
> redirect fees, and employee time) for the above m
On Wed, 27 Oct 2004 23:03:48 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
> > Anyway, considering the volume of calls you are talking about, you
> > might want to consider setting up an office supplies mailorder
&g
On Thu, 28 Oct 2004 14:00:46 -0400, Richard Branham <[EMAIL PROTECTED]> wrote:
>
> I'm attempting to set up an Asterisk server with clients as follows:
>
> SIP Client 1 (HT 286) === NAT === Internet === NAT === * Server === SIP
> Client (HT 286) 2
Double NAT ?! You may as well try to find a cure
On Thu, 28 Oct 2004 18:44:38 -0700, Jeff Rizzo
<[EMAIL PROTECTED]> wrote:
> rather this _particular_ machine. :( Are there any system tweaks I
> can try to increase the voltage (or whatever it is that's actually
> causing this)?
AFAIK, voltage can be changed in the zaptel driver.
search for t
On Thu, 28 Oct 2004 14:45:46 -0600, Ryan Courtnage <[EMAIL PROTECTED]> wrote:
> Yep, you can do this, just requires some port forwarding and special
> considerations in sip.conf.
You are missing the point. There is no *solution* to SIP NAT
traversal. All there is are *workarounds*, otherwise known
On Fri, 29 Oct 2004 07:57:03 -0600, Olger Merlos Valverde
<[EMAIL PROTECTED]> wrote:
> Ok, only one question :), this card, (X100P) looks like one modem analog :) it's
> the same?? or have some diferents...
Yes, the X100P is full-duplex, comes with a warranty that it will work
with Asterisk and Za
On Sat, 30 Oct 2004 11:12:39 +0900 (JST), Isamar Maia
<[EMAIL PROTECTED]> wrote:
> With so long distances, there is nothing better than G.729.
And why would G.729 be any better than iLBC or Speex.
In fact, if you have packet loss, G.729 absolutely sucks. iLBC is far
more forgiving and sound quali
On Sat, 30 Oct 2004 09:57:27 -0400, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> On October 29, 2004 11:49 pm, Chris A. Icide wrote:
> > Only in the X100P format, and only 2 of them
>
> I have to ask -- why are you running such high-end equipment for a craptastic
> FXO device?
The X100P is not
On Sat, 30 Oct 2004 10:59:22 -0400, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> On October 30, 2004 10:37 am, Benjamin on Asterisk Mailing Lists wrote:
> > The X100P is not only an FXO device. Many folks use it as a Zaptel
> > timing source only.
>
> Yes you are of co
On Sat, 30 Oct 2004 17:35:51 +0200, Roy Sigurd Karlsbakk
<[EMAIL PROTECTED]> wrote:
> the SIP doc on the wiki said that DTMF inband only worked on G.711. If
> I use G.726 and RFC2833 DTMF, will asterisk re-generate the DTMF at the
> Zap interface?
I'm pretty sure it does. I am using SIP phones wit
On Sat, 30 Oct 2004 14:37:35 -0600, Voip Business
<[EMAIL PROTECTED]> wrote:
> How far is IAX to be a Industry Standard I mean like SIP or H.323
Frank Miller, the author of the IAX specification document is doing
further work on the specification with the intend to eventually submit
to the IETF.
I thought this might be of interest to the list.
http://www.convergence.com.pk/iax2/trunked.html
Wasim, you should tell us about those things here. Anyway, keep up the
good work ;-)
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filter
On Sat, 30 Oct 2004 21:15:51 -0400, Steve Kann <[EMAIL PROTECTED]> wrote:
> The chart is good, but I think it makes a mistake for iLBC:
>
> Isn't iLBC 13.something kbps?
>
> Also, since iLBC uses 30ms frames (when used with asterisk, at least),
> it has slightly lower overhead. Approx 2/3 as muc
On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar
<[EMAIL PROTECTED]> wrote:
> Also does
> frequent reloads affect the stability of asterisk i mean things does it lead
> to things like memory leaks
Depends on the version of Asterisk you are using and your environment.
I have seen frequent reloa
On Thu, 21 Oct 2004 09:39:48 +0900, Benjamin on Asterisk Mailing Lists
<[EMAIL PROTECTED]> wrote:
> Wed, 20 Oct 2004 15:47:59 -0500, Henry Devito <[EMAIL PROTECTED]> wrote:
>
> > Where can I buy the act phones?
I have now discussed the matter of sample orders and shipmen
in response to many queries asking for a URL ...
http://www.voip-info.org/tiki-index.php?page=ACT%20P104SLD%20IP%20Phone
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get
On Mon, 1 Nov 2004 15:36:41 +1100, Sophus <[EMAIL PROTECTED]> wrote:
> Hi, is it possible to change the amount of time it takes asterisk to
> pickup an incoming call on a zaptel interface?
I presume you are talking about an analog FXO port here. The reason
why it takes Asterisk a while before it "
On Fri, 1 Nov 2002 09:46:46 +0300, Bilal Ghayad <[EMAIL PROTECTED]> wrote:
>
> Asterisk is working only in Linux? Can not work in Windows 2000?
You can have Asterisk on any operating system you like, as long as it
is a proper operating system that actually deserves the name, that is
to say a sys
On Mon, 01 Nov 2004 09:12:15 -0500, Tim Donahue
<[EMAIL PROTECTED]> wrote:
> But don't forget one important point, at this point there is absolutly
> NO hardware support for running Asterisk under Windows. If you need
> hardware support (TDM Cards, etc) you will need to run Asterisk on a
> Linux b
On Mon, 1 Nov 2004 08:25:03 -0600, Me <[EMAIL PROTECTED]> wrote:
> The thing is, why run it on Windows.. Even though there is a Windows version
> now it's not really a Windows version is a Linux version running on a
> version of Linux that will run on Windows.. YUCK.. That's like taking a
> Cadilla
On 01 Nov 2004 11:16:31 -0500, james <[EMAIL PROTECTED]> wrote:
> > all very easily. When I started with Asterisk, there was finally
> > something that got him interested because he's a telephone junkie.
>
> Well then, how did you expect your win-weenies to admin a hardware based
> phone system th
On Mon, 1 Nov 2004 11:06:28 -0700, public <[EMAIL PROTECTED]> wrote:
> I'm an unashamed windows user.
Nobody is. It's like alcoholics, they don't admit that they have a
drinking problem, because they can stop any time they like, right?
> So I like to game, sue me ;)
Very good point.
Leading ri
On Mon, 1 Nov 2004 16:13:15 -0600, Michael Giagnocavo
<[EMAIL PROTECTED]> wrote:
> Unless SIP just plain does not work, I think it'll be hard (for IAX to get
> excellent acceptance),
Funny you should say that from the comfort of your first world
environment. In many countries internet infrastructu
On Mon, 1 Nov 2004 16:56:27 -0600, Jay Milk <[EMAIL PROTECTED]> wrote:
> Why are you so angry?
This has nothing to do with anger, but it has all to do with truthfulness.
How many times have you had discussions with Windoze folks trying to
give you good reasongs why Windoze should have its place w
On Mon, 1 Nov 2004 17:14:37 -0600, Michael Giagnocavo
<[EMAIL PROTECTED]> wrote:
> Hi there, I'm in Guatemala. My current connectivity is via a modem (poorly
> implemented EVDO or CDMA, whichever is actually working at the moment).
> Before that we were using satellite. The telco tried to hook up A
On Mon, 1 Nov 2004 13:32:03 -0600, Jay Milk <[EMAIL PROTECTED]> wrote:
> Maybe we could keep the religious Windows vs. Linux discussions out of
> here? I think they're hardly productive, nor do they make people who
> argue for one or the other look very intelligent...
Go back and read the entire
> >Try travelling in Africa, the Middle East and South Asia, where
> >everything is still mostly dialup and many of the phone wires
> >installed go back before the time when plastic was invented.
>
> So SIP doesn't work on dialup? That's funny 'cause I'm using it like that...
That's not what I sa
[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, half protocols so to speak. IAX is a self-contained, true
interne
On Mon, 01 Nov 2004 20:55:40 -0500, Karl Brose <[EMAIL PROTECTED]> wrote:
> And what to you base such an assertion on?
> Would you care to elaborate on the technical justifications?
In a nutshell: The future of voice will be peer-to-peer and that's
where IAX has a clear edge.
As for a technical d
On Tue, 2 Nov 2004 14:25:49 +1100, Sophus <[EMAIL PROTECTED]> wrote:
> I set callerid=no in zapata.conf
>
> and
>
>exten => s,1,Wait(0)
>exten => s,2,Answer
>
> in extensions.conf
>
> but there is still a delay of about 5-8 seconds before asterisk picks up.
> any ideas?
changes in zap
On Tue, 2 Nov 2004 02:47:26 -0500, Jim Van Meggelen <[EMAIL PROTECTED]> wrote:
> You should not quote "an open-source guru" or "a guy". It undermines
> your credibility. Name the person you are quoting (Jon 'Maddog' Hall,
> President, Linux International), and provide links to prove your claim.
>
On Tue, 2 Nov 2004 10:09:41 +0100, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> > There is already one chipmaker who thought that
> > IAX was important or competitive advantage enough
> > to embed it into their chip.
>
> Which?
the PA1688
admittedly, not the best piece of silicon around, an
On Tue, 02 Nov 2004 13:50:07 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
> Out of interest, how would this work in a situation where two IAX
> compliant devices (i.e. IAXy) are behind a non-configured natted network
> (i.e. where no port forwarding has been setup)? Is it necessary to set
>
On Tue, 2 Nov 2004 17:30:20 +0800, Asterisk Mania <[EMAIL PROTECTED]> wrote:
> I would like to ask what model and brand of motherboard that can
> handle 4-6 PCI TDM04B without having a problem with IRQ etc.
A PowerMac with 6 PCI slots running Yellow Dog Linux (aka YDL).
The PM9600 (vintage model)
On Tue, 02 Nov 2004 10:42:10 -0500, Jason Becker
<[EMAIL PROTECTED]> wrote:
> You know, OpenVMS administrators would probably wet their pants laughing
> at the rhetoric coming from the Linux zealots among us.
>
> http://itmanagement.earthweb.com/erp/article.php/3380341
In a former life I used t
On Wed, 3 Nov 2004 11:57:44 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Ok, check out ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc
>
> Knock yourself out (oh, and look at the date on that file too :-) )
date?
"The requested URL /pub/asterisk/g729/unsupported/linuxppc was not
f
> > ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc
has somebody mirrored this somewhere? I never seem to have any luck
with digium's ftp site. the few times that I did manage to connect,
the connection died soon afterwards.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg
On Thu, 4 Nov 2004 10:35:02 -0500, Noah Miller <[EMAIL PROTECTED]> wrote:
> > What is a good multiline sip phone for an operator? Model and and
> > manufacturer.
ACT's P104 (10 lines)
http://www.voip-info.org/tiki-index.php?page=ACT%20P104SLD%20IP%20Phone
rgds
benjk
--
Sunrise Telephone Systems
On Fri, 05 Nov 2004 09:42:20 +0200, Altus Snyman <[EMAIL PROTECTED]> wrote:
> We have the Grandstream phone.We configured sip.conf to use libc and the
> phone as well.But now,where in alaw it worked,if you enter the voicemail
> it doesnot recognize the number you type in
set the Grandstream to use
On Fri, 5 Nov 2004 10:03:36 -, Nick Barnes <[EMAIL PROTECTED]> wrote:
>
> I asked:
>
> > I would like some form of e-mail notification to be sent when
> > a call is dropped before it's answered, or if it fails
> > through to voicemail, but is dropped before a message is
> > left. Does anybody
Also check for CANCEL, since this should be the status if the caller
hangs up before the call is picked up.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I am trying to get Asterisk to register with a SIP provider who
officially only supports ATAs of the incumbent telephone monopolist
over here.
I have so far been lucky enough to get them to ***respond*** to my
requests for information on what parameters need changing in the
REGISTER messages in or
On Sat, 06 Nov 2004 12:59:02 +0400, Vahan Yerkanian <[EMAIL PROTECTED]> wrote:
> 1.0.5.11 is the latest fully usable firmware.
Do you have the tftp server address for that? Grandstream's website
now only lists the one for 1.0.5.16.
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg
On Sat, 6 Nov 2004 06:33:15 -0800 (PST), Girish Gopinath
<[EMAIL PROTECTED]> wrote:
> AFAIK, the 050 in the From header acts as a display name. It can be used to
> determine the processing rules by other SIP entities.
[SNIP]
> The Auth credentials in the Request can be different.
Thanks,
On Sat, 06 Nov 2004 12:00:57 -0500, Karl Brose <[EMAIL PROTECTED]> wrote:
>
> The syntax for the register command is
>
> register=username:secret:[EMAIL PROTECTED]:port/extension
Trouble is though that this does not have any effect on the username
in the digest. Whatever it is intended for,
On Sat, 6 Nov 2004 16:51:44 -0600 (CST), Joe Greco <[EMAIL PROTECTED]> wrote:
> > It is real easy. EoV (ethernet over vdsl).
>
> I was under the impression that none of that stuff ran at 10Mbps or
> faster speeds. If he's got two pair and Cat3, he can just run 10Mbps
> Ethernet (and full duplex
On Sat, 06 Nov 2004 20:32:05 +0200, Gilad Ben-Yossef
<[EMAIL PROTECTED]> wrote:
>
> Well, it looks like the digest is being built with authname in
> build_reply_digest().
This seems to indicate that it was indeed the intend of the
implementor to use authname in the digest's username field and
con
On Sat, 06 Nov 2004 22:07:43 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
> I look forward to the time when you understand why your
> approach to the question was offensive.
Maybe it wasn't so much his questions ("has this been done before?"
and "where to post a bounty?") but his very unfo
On Thu, 28 Oct 2004 20:05:03 -0600, Olger Merlos Valverde
<[EMAIL PROTECTED]> wrote:
> I have two office of construction: office_1 and office_2
>
> I want connect the two offices by dedicated line, and connect the two analog
> PBX
> of this offices and transfer VoIP between two offices.
>
> Ok o
On Sun, 7 Nov 2004 10:00:25 +0900, Benjamin on Asterisk Mailing Lists
<[EMAIL PROTECTED]> wrote:
> http://bugs.digium.com/bug_view_page.php?bug_id=0002802
Mark has fixed this in the most recent CVS (Wow! that was fast!).
However, I will need this for a production system that cannot be
On Sun, 07 Nov 2004 09:47:33 +0200, Gilad Ben-Yossef
<[EMAIL PROTECTED]> wrote:
> I don't claim to understand the code at all, but what little I think I
> understand from it makes me believe this is not the change you're
> looking for.
The differences between chan_sip.c of the version on the produ
On Sun, 7 Nov 2004 01:32:18 -0700, Damon Estep
<[EMAIL PROTECTED]> wrote:
>
> If this is a must have feature, and a web interface would be an
> acceptable solution, I would look at that route. A nice benefit would be
> that you would not be locked into a particular phone set and even home
> users
On Sun, 07 Nov 2004 13:19:46 +0100, Tom Ivar Helbekkmo
<[EMAIL PROTECTED]> wrote:
> Benjamin on Asterisk Mailing Lists <[EMAIL PROTECTED]> writes:
>
> > ... and I am trying to make sense of this so as to be confident to
> > apply the change to the earlier version
On Mon, 08 Nov 2004 07:03:49 +0100, Tom Ivar Helbekkmo
<[EMAIL PROTECTED]> wrote:
> Benjamin on Asterisk Mailing Lists <[EMAIL PROTECTED]> writes:
>
> >> You found the right one. Here's what was posted to the CVS list:
> > [SNIP]
> >> Setup fromuse
On Tue, 09 Nov 2004 20:03:34 -0300, Richard Moore
<[EMAIL PROTECTED]> wrote:
> As announced before we're starting selling only E100P based
> boards. T100P boards and other "digium-like" products will
> be available in 20-30 days.
First of all, this list is not for soliciting sales.
Second, the c
Hi Walter,
I don't think it is a good idea to swamp the list with "does my modem
work with Asterisk" messages. I fear that some people are soon going
to become very angry if this trend continues.
The message you should have got from recent posts on this matter is very clear:
No modems will work
On Fri, 12 Nov 2004 14:09:55 -, James Botham
<[EMAIL PROTECTED]> wrote:
>
> Surprise, surprise you only have compatibility if you live in the US
> everybody needs to remember this...
true, but Japan is a particularly bad place when it comes to adhering
to any kind of standard. The Japanese lo
On a newly built Asterisk 1.02 system I am getting a rather strange
SIP register message ...
REGISTER sip:ispvoip-.ocn.ne.jp SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK3d1d0887
and further down ...
Contact:
Event: registration
the register directive in sip.conf looks like this:
r
sorry, I hit the send button prematurely ...
anyway, the call-ID contains the IP address of the Asterisk box ...
Call-ID: [EMAIL PROTECTED]
so it would seem the system knows it's IP address just fine.
any hints appreciated
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13
13 Nov 2004 01:47:28 +0800, Steve Underwood <[EMAIL PROTECTED]> wrote:
> The ADSI code in spandsp should handle the Japanese Caller ID, but it is
> not integrated with *.
decoding the encoded signal stream is not the problem because we can
'steal' the decoder function from Voicetronix' vpb driver
Hi again,
On Sun, 14 Nov 2004 13:06:06 -0600 (CST), Joe Greco <[EMAIL PROTECTED]> wrote:
> In this industry? Lots. Let's start with linking it to a non-GPL-
> compatible codec, move on to linking it with a propietary configuration
> and management system, and end up at creatively finding a reas
On Mon, 15 Nov 2004 17:18:14 +1300, Matt Riddell
<[EMAIL PROTECTED]> wrote:
>
> If you posted a mail saying that you thought Asterisk was crap and had
> signed this disclaimer, you could go to court. Not that Digium would do
> that though.
That's not a very good example, in my view.
Let's try
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