Thanks for sending this. However, I literally cut and pasted your
examples (with my sip credentials) and incoming calls still go
automatically to BV Voicemail. Using sip debug shows that the call never
hits my * box. Thank anyway...it was certainly worth a try.
Marios Andreou wrote:
Its worki
Thats awful funny cause you configuration doesnt work for me at all, i
even cut and pasted it as you had it and just modified the
/PPP. All i know is until today my system worked flawlessly,
so Im not sure we are getting the whole story here. i did add the three
params they suggested an
Its working just fine for me.
All IN and OUT.
sip.conf:
register => [EMAIL PROTECTED]:PP:[EMAIL PROTECTED]/
Where PPP is the password in your Account and not the login password for
BroadVoice.
is the extension to ring make sure that it is registered again with *
once you restart i
EXCUSE ME!! I changed NOTHING except added the variables you
indicated. Then incoming calls stop. So I change back to prior
sip.conf and incoming calls work again. So you tell meif they are
totally unrelated, then why do incoming calls go straight to BV
voicemail when I apply your chang
are you joking me my setup has been working fine until today when
you guys changed something on your end... dont tell me to read the
instructions after i had a working configuration that when you changed
your setup you broke, obviously im not the only asterisk user not
receiving inbound calls r
Go here...
Type budgetone in the search box on left.
http://www.voip-info.org/tiki-index.php
W
From: [EMAIL PROTECTED] on behalf of Bill Michaelson
Sent: Sat 3/5/2005 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
They are completely unrelated. Maybe you should read instructions.
Dan
On Sat, 5 Mar 2005, Mike Matthews wrote:
Why can't Broadvoice just LEAVE WELL ENOUGH ALONE!! Now, after applying
these new variables, I can't receive INCOMING calls. Sheesh, what a bunch of
BS!! Now we have to spend an
On Sat, 5 Mar 2005, Wolfgang S. Rupprecht wrote:
Does broadvoice participate in e164.{arpa,org,info}?
Yes
Does this change mean that non-customers can't call broadvoice
customers with a pure SIP call by routing the call to
sip.broadvoice.com?
Calls can be made to broadvoice phones by @sip.broadvoi
Jim Van Meggelen wrote:
I would like to start a discussion centred around the various ways one
might serve up configuration files from an Asterisk server (I know, it's
better to use a secondary server for all this, but let's talk about a
smaller system).
The types of things being served would inclu
Hi John,
Here’s the best place for you to
start looking for voip providers
http://www.voip-info.org/wiki-VOIP+Service+Providers+Business
and no I work 24x7.
Trolled meant it is such a basic question that’s
all but hey everyone has to start somewhere once.
Cheers,
Dean
Are you guys crazy... no change control, no customer notification, no
inbound services now work, what the hell kind of business are you
running there. As a Broadvoice user, soon to be X-Broadvoice user I have
no idea how your going to stay in business doing this kind of things to
customers. Is this
[EMAIL PROTECTED] (Dan Weber) writes:
> I was doing the best I could to get the situation under control. This
> really was out of my hands, but I'm trying to repair as fast I could.
> If you haven't received the email regarding the change yet, please
> notify me.
Does broadvoice participate in e
Debian leaves the burden of choice to the user. It carries 3 tftpd-s, at
least 3 dhcpd-s and at least 16 httpd-s
Another note regarding memory usage:
Apache2 offers better memory usage with the thread pool model (rather
than the process pool of apache1). But just remember that mod_php4 still
doesn
Why can't Broadvoice just LEAVE WELL ENOUGH ALONE!! Now, after
applying these new variables, I can't receive INCOMING calls. Sheesh,
what a bunch of BS!! Now we have to spend another weekend fixing what
BV screws up.
Dan Weber wrote:
Today, We have added INVITE Authentication. This seem
On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote:
> Hi,
>
> I am having some trouble installing asterisk addons on Debian. I wish to do
> this to use mysql billing.
>
> I have mysql and mysql-devel packages installed I think!?
Yes, it should do (together with a build environment).
I
Hi,
My termination with sixtel stopped working, is it something I did or anybody
else is having the same problem.
I am attaching log:
*CLI>
-- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack
-- Goto (macro-dialout-default,s,4)
-- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack
On Sat, 5 Mar 2005, Daryll Strauss wrote:
Finally, posting on the Asterisk list to tell people you broke your
system isn't sufficient. What if the user isn't reading
asterisk-users. Heaven knows it's tough to keep up with this list. You
really need to reverse this change, notify all your customers
I was doing the best I could to get the situation under control. This
really was out of my hands, but I'm trying to repair as fast I could. If
you haven't received the email regarding the change yet, please notify me.
Dan
On Sat, 5 Mar 2005, Jerry Glomph Black wrote:
Thanks for this info, Dan!
This is out of my hands, I'm sorry. I'm just trying to make it fixed as
fast as I can.
Dan
On Sat, 5 Mar 2005, Brian Roy wrote:
On Sat, 5 Mar 2005 12:13:08 -0500 (EST), Dan Weber <[EMAIL PROTECTED]> wrote:
Today, We have added INVITE Authentication.
Thanks for the warning. You pissed my wife off
On Sat, 5 Mar 2005 12:13:08 -0500 (EST), Dan Weber <[EMAIL PROTECTED]> wrote:
> Today, We have added INVITE Authentication.
Thanks for the warning. You pissed my wife off. If she can't make
calls, she's an unhappy camper. Maybe next time you warn us?
Sheesh
-Chuji
_
On March 5, 2005 08:00 pm, Jim Van Meggelen wrote:
> I have heard that khttpd is pretty lightweight, but its use seems to
> have been deprecated, and it does not appear to be actively maintained.
> Is TuX the way to go?
>
> As for tftpd and ftpd, I'm just not sure. Leightweight is the key, here.
>
> I would like to start a discussion centred around the various ways one
> might serve up configuration files from an Asterisk server (I know, it's
> better to use a secondary server for all this, but let's talk about a
> smaller system).
>
> The types of things being served would include:
> - Log
Jim Van Meggelen wrote:
I would like to start a discussion centred around the various ways one
might serve up configuration files from an Asterisk server (I know, it's
[snippage]
I have heard that khttpd is pretty lightweight, but its use seems to
have been deprecated, and it does not appear to
http://www.grandstream.com/user_manuals/budgetone100.pdf
Mike Chapman wrote:
Hi-
I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to
make sure it has dhcp enabled and it is. The display says no
I would like to start a discussion centred around the various ways one
might serve up configuration files from an Asterisk server (I know, it's
better to use a secondary server for all this, but let's talk about a
smaller system).
The types of things being served would include:
- Logo image for se
Not that I need to stick up for Broadvoice and yes, they are not very
good at returning emails, but, for me, I have used Broadvoice on several
asterisk systems at different locations and haven't had any problems.
It works great and is very flexible.
-Original Message-
From: Daryll Strauss
I really don't like speaking about it, since it's a topic that will
never go away on it's own if we don't speak about it, nor will it go
away if we do speak about it. Remember yesterdays wannabees are todays
newbies, and todays newbies are tomorrows experts, and so on. The
newbies that see this thr
We are a medical foundation and a voip indial service
is a possiblitiy. I wasn't aware that such a thing exists. The
person who proposed this to us would have probably explained it all to me if it
were not the weekend. He is a volunteer for the foundation.
Side question: You feel you ar
> I think it comes down to tolerances. Some days I am far less tolerant than
> others when it comes to people who just don't want to put forward any effort.
>
> Today was particularly bad and if I was a smarter person, would probably
> refrain from reading the list at all. I don't always sel
Is it possible to use the Hold/Transfer/Conference/Flash keys of the
Budgetone-101 (FW 1.0.5.22) with Asterisk?
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Sounds like you are having a codec issue with 2 of your providers.
Make sure you find out what codecs are supported and that your config
is set up accordingly.
On Sun, 06 Mar 2005 00:14:05 +, w fm3 <[EMAIL PROTECTED]> wrote:
> Hi
>
> Hope someone can help :)
>
> I am testing 4 PSTN termin
> Has anyone else experienced this. We are setting up ethereal to monitor
> things to collect more information but are hoping others might be able
> to shed some light on this.
Yup George
I get a similar issue by rebooting the pbx, and not having it register again
+ the busted redial button, & xfe
Nigel,
Should really be on the biz list for this, but Telappliant sells Digium
hardware.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nigel
Taylor
Sent: 05 March 2005 21:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asteri
Hi
Hope someone can help :)
I am testing 4 PSTN termination providers. 3 SIP and 1 IAX
IAX and 1 of the SIP providers work fine.
Now the wierdness:
2 SIP providers I can only get oubound calls to ring at the destination and
then nothing more. 1 gets as far as SIP code 183 (and ringing on the src
On Sat, 5 Mar 2005 15:02:47 -0700, Gabriel Gunderson <[EMAIL PROTECTED]> wrote:
> May I suggest:
>
> 1) Updating your website that tells how to configure Asterisk for Broadvoice.
>
> 2) Answering emails to [EMAIL PROTECTED]
>
> 3) Emailing your users that signed up as BYOB when you think a chan
Hi there,
I read from the mailing list that people is using some patches to do
special things like fax support (or something related) and other stuff
that seem very useful.
Like the spandsp patch for * fax located at http://www.soft-switch.org/
Also, www.voip.info.org shows hundreds of good links
Hi Chetan,
Per my understanding to the chan_h323 operation, if no codecs are loaded,
Asterisk will perform a pass-through function, which means that signaling is
passed via Asterisk, but RTP passes between the endpoints.
This is NOT proxy function, as with proxy function means that RTP passes
George Pajari wrote:
The Netscreen monitors the REGISTER messages and only keeps the reverse
mapping open for the duration of the registration period. It appears
that every so often the Sayson does not send out another REGISTER
message after the registration has expired resulting in the reverse
map
Gabe / Jerry,
I agree with most your points. The BYOD plan and asterisk
compatibility is the biggest selling point for me too.
I want to point out, however, that the info in the Wiki
(http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice) already
shows the use of username/fromuser/secret. I
[EMAIL PROTECTED] (Gabriel Gunderson) writes:
> As an "early adopter" kinda guy, I'm happy to tweak stuff to make
> things work. I can't however explain to my wife why the phone doesn't
> work *again*. I'm going to hang in there a bit longer in hopes that
> things will get better, if they don't,
> > My feeling (unsupported) is that the powercycle does a better job of
> > forcing the far end
> > of an E1 (e.g. the PTT's equipment) to start afresh than just
> > reinitializing the cards.
> > If you turn the power off you can be sure that you are going to drop
> > carrier, clock and any
> > co
We have a customer with a handful of Sayson/Aastra 480i phones behind a
Juniper Networks Netscreen firewall registering with our hosted PBX service.
The Netscreen monitors the REGISTER messages and only keeps the reverse
mapping open for the duration of the registration period. It appears
that ever
> Most of those people who tend to scold every newbie are probably
> "elitist pretenders". If they were truly elite, they would be too busy
> to read and reply to such posts. Of course some of us are just
> impatient, regardless of our skill level.
Very well said!
Now back to helping those th
Andrew Kohlsmith wrote:
I can't tell if you're on "my side" of this debate or not, because you sure
sound like it. :-)
I'm on "my side" and always on the side of the serious and earnest end
user. I live in a sparsely populated rural area. Most of the locals
can't afford to pay for computer
On March 5, 2005 04:49 pm, Paul Fielding wrote:
> Geepers man. Looking at the last couple of times you tried to 'educate'
> someone, I don't see anything in their messages that sound like "I need X
> done **RIGHT NOW**!! I DEMAND HELP!!". Uneducated and demanding are not
> necessarily the same t
On March 5, 2005 04:57 pm, Paul wrote:
> FWIW - some of us have paying customers and helping them takes absolute
> over helping newbies who won't try to help themselves first. Some of
> them actually are capable of reading and studying the information
> already out there, but they are also selfish
Have you readed that page, because I do a reseach (before I put the mail to
the list) and I found nothing.
LTenorio
-Original Message-
From: Robert Webb [mailto:[EMAIL PROTECTED] On Behalf Of Robert Webb
Sent: Saturday, March 05, 2005 7:26 PM
To: Asterisk Users Mailing List - Non-Commerc
> -Original Message-
> From: Robert Webb [mailto:[EMAIL PROTECTED]
> Sent: Saturday, March 05, 2005 5:24 PM
> To: 'Asterisk Users Mailing List - Non-Commercial
> Discussion'; 'leandro_tenorio'
> Subject: RE: [Asterisk-Users] IAX2 (Variables)
>
>
>
> > -Original Message-
> > From:
Seems sometimes.. Specially with certain iax softphones like iaxcomm...
Seems firefly doesnt have any problems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: Sábado, 05 de Marzo de 2005 03:56 p.m.
To: Asterisk Users Mailing List - N
Anyone knows what are the variables in an inbound IAX2 call who
reflect the actual codec and DNID, DNIS, original peer description, I'm only
able to see it during an iax debug
Timestamp: 3ms SCall: 1 DCall: 0 [XX.XX.XX.XX:5036]
VERSION : 2
CALLED NUMBER : XX
An Open letter to Broadvoice from an Asterisk user...
(This is not a solicitation for support from the Asterisk list. The
specifics of my problems have already been emailed to their support
team.)
May I suggest:
1) Updating your website that tells how to configure Asterisk for Broadvoice.
2) A
Can anyone recommend a source of Digium hardware in the UK ?
Thanks in advance
Nigel
begin:vcard
fn:Nigel Taylor
n:Taylor;Nigel
org:ITAzure Limited
adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom
email;internet:[EMAIL PROTECTED]
title:Technology Director
tel;work
Andrew Kohlsmith wrote:
On March 5, 2005 02:46 pm, Paul wrote:
Maybe one of the free web-based forum packages will eventually offer an
"elitist" or "impatient" mode. Before you can post, you do the required
reading and pass online exams. The idea is to weed out people who think
README is just an
Fredrik wrote:
> I see from my CDR's that some of my callers also have "unknown" in
> their FROM field. I would like to let them through. Only block the
> FROM "anonymous" that the telemarketers use.
Fredrik, I found something on the Wiki a while back... Try this...
exten => s,1,Answer
exten => s,2
I don't know if this is still true, but Iax clients had problems when you
check them with qualify (set latter to no)...
HTH,
Rob.
- Original Message -
From: "Anton Krall" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Saturday, March 05, 2005
Thanks for this info, Dan! I noticed immediately that outbound was broken, and
inbound was OK. I saw your posting just prior to going berserk... a warning
email from Broadvoice would have been nice, they knew to email me when that
SIP-patch from edvina came out some months ago.
Anyway, thank
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
I think where the problem comes in is that people take this forum to be
asterisk-biz half the time. "I need X done **RIGHT NOW**!! I DEMAND
HELP!!"
-- take it to -biz, there are dozens if not hundreds of consultants who
Can anyone recommend a Digium Reseller in the UK ?
Thanks in Advance
Nigel
begin:vcard
fn:Nigel Taylor
n:Taylor;Nigel
org:ITAzure Limited
adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom
email;internet:[EMAIL PROTECTED]
title:Technology Director
tel;work:0116 286
Does any have ... or know where I can find firmware to convert a
DVG-1120M (MGCP) to a DVG-1120S (SIP)??
Thanks,
Rob
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> As far as I can make out the root password for the ISO download is
> supposed to be epping or EPPING depending upon which version you are
> using.
> I've downloaded an ISO image from the following link but neither passwords
> seem to work :(
>
http://ovh.dl.sourceforge.net:80/sourceforge/asteri
On March 5, 2005 04:15 pm, tim panton wrote:
> My feeling (unsupported) is that the powercycle does a better job of
> forcing the far end
> of an E1 (e.g. the PTT's equipment) to start afresh than just
> reinitializing the cards.
> If you turn the power off you can be sure that you are going to dro
Anyone using this Sip phone with Asterisk?
If you have had success getting the message waiting indication to work,
please contact me off list.
TIA
John Novack
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On 5 Mar 2005, at 18:44, Alfredo Sola wrote:
Hi,
I've no experience with the TE110, but this is a known problem with
the TE405 and TE410. They apparently can get locked up, and only a
power cycle will clear it.
Good hint, I'll take that into account when testing.
Hope the TE100 is bett
Well, considering I'm on topic, I shouldn't get flamed to badly for this. I
have a bunch of these working well in my home experiments:
http://www.laptops4me.com/product_info.php/products_id/1444
And yes that price is correct and they do arrive. :)
Not everyone can justify buying the "supported"
On March 5, 2005 02:46 pm, Paul wrote:
> Maybe one of the free web-based forum packages will eventually offer an
> "elitist" or "impatient" mode. Before you can post, you do the required
> reading and pass online exams. The idea is to weed out people who think
> README is just another geek buzzword
On March 5, 2005 02:30 pm, Mike Dent wrote:
> There is a difference when you are that childs father or mother but
> you are neither?
You have a point, but... (read on)
> Do you stop people in the street when you see them doing things wrong
> and try and tell them "you shouldn't be doing it like t
Hi list!
I'm using phones that emulate a Cisco 7940 with chan_sccp. When I was
using Asterisk 1.0.5 (bristuffed) I never had any such message on the
console.
The phones do work.
Is this a bug in chan_sccp or a feature of asterisk 1.0.6?
Thx!
___
Asteri
Hello! I am a newbie with asterisk, I´d like to install capi on FC3, I´ve
tried to follow a little howto
(http://voip-info.org/wiki-Asterisk+Linux+Fedora), but it is for FC1, and
when I do a modprobe fcpci it fails (module not found).
Please some help!!
_
What iax2 softphones are you guys using?
Ive trying some but I find some lack certain features and others have them
but lacks others.
For example, I tried firefly, simple interface but seems it can only handle
1 line, no MWI.
IAX Phone has multiple lines and MWI but seems it can only handle gsm
Mike Dent wrote:
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCop
There is a difference when you are that childs father
On Sat, 2005-03-05 at 19:30 +, Mike Dent wrote:
> >
> > And if nobody's going to educate the newbies, then how will they ever learn?
> > Do you believe in letting your children do whatever they want, too? There
> > are 'defacto' rules for any system. No, I don't have my shiny ListCop
>
> Th
Dnia 2005-03-05 15:04, Użytkownik Marcin Zajączkowski napisał:
I've just compiled and installed Asterisk (1.0.5). After some problems
with codecs I could successfully connect to server by:
[EMAIL PROTECTED]
Next I created account at iaxtel.com and configured iaxcomm to work with
this account. Un
Hi Nathan,
Nathan C. Smith wrote:
I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee
phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the
Try the wiki, myself and someone else wrote up a pretty big howto and
tips and tricks on these phones.
http://voip-in
>
> And if nobody's going to educate the newbies, then how will they ever learn?
> Do you believe in letting your children do whatever they want, too? There
> are 'defacto' rules for any system. No, I don't have my shiny ListCop
There is a difference when you are that childs father or mother bu
- Original Message -
There is a LOT of traffic on this list about products that are not
supplied by Digium. Do you want to exclude those also?
The Sangoma guys typically handle support for their own product, even on
this
list. Atacomm's card hasn't hit the market yet. The Sipura people
Hi!
> I'm looking into solutions for providing a live stream of an event in
> Belgium [1] - for example, as follows:
How about icecast:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices
Another approach:
Dial into a MeetMe conference, and connect some client to that conference
On March 5, 2005 12:02 pm, John Novack wrote:
> Since Digium no longer suppliers this card, they were denied NOTHING!
They offer comparable hardware. TDM410P is $113.
> There is a LOT of traffic on this list about products that are not
> supplied by Digium. Do you want to exclude those also?
Th
Hi,
I've no experience with the TE110, but this is a known problem with the
TE405 and TE410. They apparently can get locked up, and only a power
cycle will clear it.
Good hint, I'll take that into account when testing.
Hope the TE100 is better built than that, though. At least, for the
On March 5, 2005 11:57 am, Dave Cotton wrote:
> Just a question, where's the Dev Lite Kit on Digium's site?
I meant "Dev Kit" and you're right, it's an FXO and FXS on the carrier card.
US$195.
> The PCI Dev kit would give him an FXO and an FXS which may be more than
> some people want, perhaps
The code is configured to allow use of either mysql or postgres, so you
will
need to install the postgres-dev package, or comment out all postgres
related
code.
Once you have the postgres libraries installed you have two more changes
to make.
line 645 needs to become:
AST_MUTEX_DEFINE_ST
Hi,
I have two Asterisk servers and I forward calls from one to the other.
How do I reload extensions included in a switch statement in
extensions.con? I have tried "extensions reload", "reload" and "restart
now", and it's only "restart now" that works. Is this how it is supposed
to work or can
On March 5, 2005 11:01 am, Andrew Kohlsmith wrote:
> specifically these types of problems. Maybe someone else on this list is
> more forgiving than I am but I really hope not.
I apologize for this remark. I still do feel, though, that if you're this new
to asterisk that you should have purchase
On March 5, 2005 12:22 pm, Rich Adamson wrote:
> There seems to be about a half dozen "self-appointed" list cops, and
> none of them speak for Mark, digium or asterisk. Several of those are
> lurking on this list only to find fresh meat to sell their services to.
> It's obvious who they are.
I hav
Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057
Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is already
loaded):
Not to
That worked a treat many thanks.
--
Regards
Phil
--
This message was scanned for spam and viruses by BitDefender.
For more information please visit http://linux.bitdefender.com/
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> I've downloaded an ISO image from the following link but
> neither passwords
> seem to work :(
>
> http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/as
> teriskathome-0.6.iso
>
> any one know the password for this one?
>
> --
> Regards
>
> Phil
>
The root password for 0.6 is
Rich Adamson wrote:
There seems to be about a half dozen "self-appointed" list cops, and none of
them speak for Mark, digium or asterisk. Several of those are lurking on this list only
to find fresh meat to sell their services to.
It's obvious who they are.
Indeed.
Much easier to config the
Great, thanks, that was the information I was looking
for.
--- Rich Adamson <[EMAIL PROTECTED]> wrote:
> > Has anyone done Voice Over Frame Relay with
> Asterisk.
> > With Frame Relay work reliably with Asterisk? Any
> > experiences?
>
> If you're talking about transporting voip calls
> acros
> Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My
> understanding (someone please correct me if I'm wrong) is that this list *is
> not* a Digium support list. This list is a forum for Asterisk discussion by
> users. As such, I would suggest that all topics of discus
As far as I can make out the root password for the ISO download is
supposed to be epping or EPPING depending upon which version you are
using.
I've downloaded an ISO image from the following link but neither passwords
seem to work :(
http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/ast
Something like this sis similar to what you are looking for I think.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felix E. Klee
Sent: 05 March 2005 17:17
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk for Live-Stream?
I'm loo
> Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL
> with 640kbits
upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll
change it soon
for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X.
>
> Is there another cod
Thank you Wiley.
I guess I had the "q" version installed. I
removed everything and tried with "r" and followed the wiki instructions. Simple
Answer-MusicOnHold works fine, so I guess my problem is resolved.
- Original Message -
From:
Wiley
Siler
To: Asterisk Users Mail
I'm looking into solutions for providing a live stream of an event in
Belgium [1] - for example, as follows:
* Event --> mobile phone --> software answering machine --> Internet
server
* Event --> mobile phone --> VOIP --> Internet server
The live stream should be available in a format so that
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My
understanding (someone please correct me if I'm wrong) is that this list *is
not* a Digium support list. This list is a forum for Asterisk discussion by
users. As such, I would suggest that all topics of discussion for
> Has anyone done Voice Over Frame Relay with Asterisk.
> With Frame Relay work reliably with Asterisk? Any
> experiences?
If you're talking about transporting voip calls across a path that
includes frame relay links, yes it works just fine "if" you frame
network is not congested.
Frame relay n
Today, We have added INVITE Authentication. This seems to bring a large
amount of problems to people in the way since they can't make outbound
calls. Here's what needs to be done. You need to add three variables to
your peers or friends, username, authuser, and secret.
username=
authuser=
se
Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL with 640kbits upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll change it soon for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X.
Is there another codec that do t
Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057 Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is
On Sat, 2005-03-05 at 11:01 -0500, Andrew Kohlsmith wrote:
> On March 5, 2005 08:14 am, Androtech wrote:
> > I bought one "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057
> > Motorola Chip, and I installed it on my linux box.
>
> > When I try to load the module wcfxo, I cannot load it
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