On 9/17/07, Dan Austin [EMAIL PROTECTED] wrote:
Lacy's response in the thread 'Why does
everyone seem to dislike *now?', has a small
bit that caught my eye.
Chan_Skinny made a lot of progress between 1.2 and
1.4, and even more in the later 1.4.X releases.
I am curious as to which
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote:
Greetings,
Last week I began researching Asterisk for the first time. I did what most
noobs would do; downloaded an image that seemed simple and straightforward
and had some credibility (*now). I also downloaded the TFOT version 1 as a
On 9/13/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Hi, I have a production asterisk-1.2.8 system with FreePBX PRI Digium
card.
I am looking for a paging system to an external speaker. I can page to
internal Polycom 501 VoIP.
But, what hardware or system do I need to integrate with the
On 9/7/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
We have users with Cisco 7900 phones running sip. When user A calls
user B, we want user B's name to appear on user A's phone. It shows the
extension they call, but not the internal name of the called user. Is
this possible? We
On 9/4/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do you know where to find clear developers' guides (with some
examples)
for developing apps that run *on* Cisco 79xx phones (especially the
7970)? Examples that can run against Asterisk (not CallManager) with SIP
firmware (not
On 9/1/07, Dovid B [EMAIL PROTECTED] wrote:
Why work with two separate devices when you can have one ? And yes the DC
is
staffed 24/7 but do you want to call them every time you need a new CD/DVD
inserted in to the box when you are working on it ? IMHO A rac card + a
better server is worth
On 9/2/07, Nick Adams [EMAIL PROTECTED] wrote:
Lacy Moore - Aspendora wrote:
On 9/1/07, *Dovid B* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Why work with two separate devices when you can have one ? And yes
the DC is
staffed 24/7 but do you want to call them
On 8/16/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
This is really ridiculous. So this means that now nobody can use
fax-to-email without paying to J2 first?
Welcome to America! Home of the Ridiculous! The crazier and ridiculous it
is, the more it's likely to be true.
But, seriously, I
On 8/10/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I am having a bit of a problem implementing the pickup command in
my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and
I
can get the
More specifically:
https://www.digium.com/en/wheretobuy/digiumdirect/voice_prompt.php
On 8/9/07, Cory Andrews [EMAIL PROTECTED] wrote:
linky
http://www.digium.com/en/products/voice/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent:
On 7/6/07, Stephen Bosch [EMAIL PROTECTED] wrote:
The price of open source is that the commercial outfits are free to rip
off ideas without paying for them.
And then patent those ideas and call them their own, knowing full well open
source developers can't hire the attorneys necessary to
On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote:
Contrary to the opinions of Anglo-Philes, we, here in the Colonies,
speak American, not English. In some places, 'Murican.
We get to do that, because, back in the late 1700's . . . we won.
It is only referred to as English out of a sense of
On 7/3/07, J. Oquendo [EMAIL PROTECTED] wrote:
You're answering your own question. Forwarding a call with a number
that is not the originating number is what (drum roll)
And in a corporate environment, what is the originating number? Is it
the main line, the DID, or what?
If I am at my house,
On 7/3/07, Farooq Ahmed [EMAIL PROTECTED] wrote:
Hi all,
As we know we can configure in astersik like before 5:00pm calls go to
reception and after 5:00
pm calls go to some mobile no. One of my client requested that he wants to
manually shift the dial
plan like above as he has flexiable
On 7/3/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
And frankly, *NO*... I don't want to give anyone my cell number. Once
you give out the cell number, people call you on it before they attempt any
other number.
You are absolutely correct. I walk down the hall of our office and see
On 6/29/07, Ade Vickers [EMAIL PROTECTED] wrote:
What I'd like to do is have the music streaming constantly, so the on hold
caller always gets music at the current position; even if that's in the
middle or near the end of a file.
Many of us would like this, but the powers that be decided they
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
Thoughts vary to second T1, with channel bank, breaking out some DS0's into
a channel bank, or finding a T1/fax board (do they exist?), to go directly
into the FAX server (PC/linux based)
It looks to me like you have two choices. The first
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
One idea is to utilize DID, and have Asterisk forward the calls to the
current FAX lines, preserving the DID as Caller ID. I am fairly sure
Asterisk itself can do this. (The call would appear to be from this
assigned ID). If so, I could,
On 5/24/07, Paul Aviles [EMAIL PROTECTED] wrote:
is there a way to support login and logout functionality in a phone? We are
using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use
the same phone using like 2 different extensions. The phone will then
remember your settings
I like to forward them back to themselves, that is, the ones that give
their phone number. Check nerdvittles.com. I think he had some kind
of torture script setup, if I remember correctly.
On 5/5/07, Salvatore Giudice [EMAIL PROTECTED] wrote:
Just forward them to 1-800-big-dick or some other
] wrote:
On Sat, 2007-04-14 at 14:17 -0500, Lacy Moore - Aspendora wrote:
This was mentioned earlier:
I suspect IRQ Sharing.
I know. And I posted my /proc/interrupts showing that there were no
shared IRQ's.
And from the rest, it sounds like your network card and Digium card
are both
On 4/14/07, Greg Woods [EMAIL PROTECTED] wrote:
Even worse, I discovered
that the same problem affects racoon/ipsec-tools as well; I get racoon
errors in the log about hash mismatches and messages too short. Unload
the zaptel drivers, and the tunnel is established immediately. I was
hoping to
On 4/14/07, Greg Woods [EMAIL PROTECTED] wrote:
On a possibly related note, I find that I cannot build the Zaptel
drivers at all on newer FC6 kernels. I am running 2.6.19-1.2911.6.5.fc6
Never mind about my previous message about compiling Zaptel. It's
unrelated, but what may be related is
You might try doing a database lookup, but you'll still have to enter
all 200 caller ids by hand. I think the database lookup would
probably be better than adding 200+ extra lines to your dial plan.
There's probably something in AEL that you could write that might be
more effecient.
On
I will add one thing. Parking might be a little problematic out of
the box. If you don't have problems using a patch that is not in the
main branch, there is a valet parking patch that would handle this
without any problems.
On the other hand, if the companies do not have to be 100% separate,
On 4/13/07, Patrick [EMAIL PROTECTED] wrote:
Do you know where this patch can be found? My googling came up empty.
http://www.freeswitch.org/asterisk_stuff/
app_valetparking.c works on 1.4. You have to add it to the menuselect
file. There's also a version for 1.2. I'm using it with 1.4
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
exten = 104,Dial(SCCP/SEP00036BC3852B,20)
exten = 104,2,Voicemail(u104)
exten = 104,102,Voicemail(b104)
exten = 104,103,Hangup()
Off the top of my head, I would say that your dial statement should be
Dial(SCCP/104,20). You should be
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
exten = 104,Dial(SCCP/SEP00036BC3852B,20)
exten = 104,2,Voicemail(u104)
exten = 104,102,Voicemail(b104)
exten = 104,103,Hangup()
Actually, if this is a cut and paste, you are missing the 1. It should be:
exten = 104,1,Dial...
you have
On 4/10/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
chan_sccp with the patches for 1.4
Everything should be fine, then, unless maybe you have really old
firmware on the phones. That's the only thing I can think of. I've
been running 1.4.2 with chan_sccp for a while in a test environment
On 4/6/07, Steve Prior [EMAIL PROTECTED] wrote:
I just found out that the celldock I'm talking about is also called the
Dock-N-Talk.
Works just fine. There is a delay, actually a LONG delay from the
time you dial the number and the cellphone connects the call. Or, at
least with my Motorola
We should have a welcome back to work party for fb.
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On 3/29/07, Brad Stockdale [EMAIL PROTECTED] wrote:
Hello all,
loadInformation7 model=IP Phone 7960P003-08-6-00/loadInformation7
Should be POS03-08-6-00. The same as your .loads file. Also change
this in the OS79XX file.
P003-08-6-00.bin
P003-08-6-00.sbn
P0S3-08-6-00.loads
On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I am using autoload and I have rebooted the server. I have tried using
different files and a different location. This is getting very
frustrating.
I wish I knew what the problem was.
Not that it will help me, because I'm pretty much
I just noticed the Aastra 57i do something that I haven't seen before.
I called from one phone (phone 1) to the 57i. I answered it. Then,
I pressed Transfer and dialed the extension for the third phone (in
this case a Cisco 7960 in Sip). I did not answer the Cisco, but
noticed the caller ID
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to describe it, seems like the voice
packets are being played out too fast.
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Hmm... Not sure. But this really sounds like ztdummy is not working
correctly. Hopefully someone else can jump in here. The only system
I've ever done without a
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
ztdummy 4424 0
rtc11156 1 ztdummy
zaptel178084 1 ztdummy
crc_ccitt 2016 1 zaptel
Ok, this is a dumb question, but what is that output from?
What distribution of Linux are you
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium
cards. The problem I have is that MOH will not play. It starts and then
stops.
If you rub your hand across the mouthpiece of the phone, does the music play?
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
WOW that fixed it! What an Idiot.
I was going somewhere with that, but never mind. Good luck.
Maybe the idiot is the guy who posted no additional details of his
configuration, in particular, whether the CLI was showing music on
hold
On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
The cli shows:
-- Started music on hold, class 'jessica', on channel 'IAX2/205-3'
-- Stopped music on hold on IAX2/205-3
That rules out the timing.
I see this note in the config file:
; If you are not using autoload in
On 3/21/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Hi all,
I have just successfully configured a Cisco 30VIP to work with my
Asterisk server. I have seven of these phones new and would like to
deploy them. I am wondering if anyone has this phone deployed with
Asterisk and can suggest
On 3/22/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
1.4.1
I've got one of those at home and a test system running 1.4.2. I'll
take a look tonight and see if there is anything obvious. I'm not a
developer, though. I know one of the guys working on chan_skinny uses
30VIPs, so I would have
On 3/22/07, LKS GMAIL [EMAIL PROTECTED] wrote:
Yeah, I know but the problem begins when i try to pick a call up from IAX or
ZAP not in SIP.
Again, read the documentation at www.voip-info.org, specifically . It
is possible with Zap, I'm doing it. Granted, I'm not doing it with
IAX, but I am
On 3/22/07, dave cantera [EMAIL PROTECTED] wrote:
thomas,
the dialplan is quite different in 1.4.x... they use a users.conf file for,
I think, all endpoints (phones not providers)... there is no documentation,
Somebody forgot to tell me I had to use the users.conf file. Hmmm
Guess
On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
First of all, thanks a lot.
Believe me that if I'm writing down here it's due that i cannot find the
problem out. Maybe it's a bug, but either of IAX or mISDN couldn't get
pickup calls.
Could be the GrandStream?
Forgive my lack of knowledge on this,
On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
Could be the GrandStream?
What's the firmware version and hardware version (if the hardware is
v2, it will say v2.0 on the back of the phone)?
You may be using older firmware that doesn't support the pickup. I
have no idea when it was added.
On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
It's no necessary to use BRIstuff for this issue... but i'll try!
No, I didn't mean to try BRIstuff.
What version of Asterisk? This is strange.
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On 3/22/07, Lukas [EMAIL PROTECTED] wrote:
It's so strange... i don't know what's happens.
Looks like _I'm_ the one that needs to read the documentation. I
thought you could pick up a ringing Zaptel channel. But, I couldn't
get it to work on my 1.4.2 system.
I could pick up the Sip device
On 3/19/07, Scott Plante [EMAIL PROTECTED] wrote:
work better in general. Is it the general experience on the list that
SIP is more mature and reliable than IAX? We like the fact that we don't
have to open inbound ranges of ports for IAX to work. We are in Atlanta
I've switched to using SIP on
On 3/14/07, Steve Totaro [EMAIL PROTECTED] wrote:
Just an FYI in case you didn't know, there is also a callcenter asterisk
mailing list that you could post this to. I am not sure how many users
are subscribed but it is most certainly more of your target audience.
Where do you subscribe to
On 3/14/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
The third field, in my case Local/4${BRIDGEPEER:5:[EMAIL PROTECTED]
is the channel to announce the parked call slot to. In my case,
extensions beginning with 1xx are the phones themselves, and extensions
4xx are the same phones
On 3/10/07, Henry Cobb [EMAIL PROTECTED] wrote:
So get a second broadband connection and run only voice on it.
Has anyone tried this?
I have been thinking about this. We're getting so much spam that I
think it's taking up too much of our bandwidth. I'm wondering how
much bandwidth all the
We're not running echo cancelling cards here. We may have 1 or 2
phone calls a month with echo, and it's primarily calls to a certain
number. When asked about the echo, I explained the difference in
price, and for the price difference, we can deal with the echos.
For the most part, for us,
On 3/6/07, Russell Bryant [EMAIL PROTECTED] wrote:
This will connect the
station to the first available trunk if there is one, and then provide
dialtone for making a call.
That's what I was concerned about. Whether it connects to the first
available, or the first one. In other words, if
On 3/3/07, Mike D'Ambrogia [EMAIL PROTECTED] wrote:
Wanting to connect my asterisk box off of 2 unused analog extensions on the
non* PBX system.
Sounds workable.
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On 3/2/07, Russell Bryant [EMAIL PROTECTED] wrote:
If you are interested in beginning to look at it now, just pull the code
from the 1.4 branch.
Russell, I don't have any specifics at this time. I need to dig a
little further. I'm thinking the autocontext is what is giving me
fits. I can
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
My point is that if it's going to involve rebuilding a kernel to support
IO-APIC, then I'd just as soon build from the ground up.
And my point is that this is the Asterisk Users mail list, not the
Trixbox list. Either ask other there or ask
On 2/22/07, Frederico Madeira [EMAIL PROTECTED] wrote:
My asterisk is show me some errors on line registration.
This message appear on console: Request to schedule in the past?!?!
I could be wrong here, but I think one of the symptoms of that could
be not have any zaptel devices and not having
On 2/22/07, Norbert Zawodsky [EMAIL PROTECTED] wrote:
Does someone know if it is possible to light up a LED under this szenario?
1.2 or 1.4?
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On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
Does Trixbox support
www.trixbox.org
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On 2/14/07, George Wise [EMAIL PROTECTED] wrote:
Does anyone know of a good Asterisk/LAN/PC support company in Houston, TX?
Yep
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I'm wondering if anyone else has experienced this. Up until a few days ago,
when accessing the CLI from my terminal program (Private Shell), the output
was in color. I haven't upgraded, rebuilt, or to my knowledge, changed
anything in Asterisk that would change this. My terminal settings were
On 2/12/07, Bruce Reeves [EMAIL PROTECTED] wrote:
I have seen this when I have restarted the server from the asterisk CLI
and not a service asterisk restart command. I'm not sure as to why, but I
always assumed it had to do with the safe_asterisk file.
Bruce, that may have been it. I just
On 2/8/07, Remzi Semsettin Turer [EMAIL PROTECTED] wrote:
This is a solution if your provider is using IAX, but we are stuck with
SIP.
Huh? What do the two have to do with each other?
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On 2/3/07, Jim Karen Ostrosky [EMAIL PROTECTED] wrote:
Hi, first time poster. I've searched, but find very little on this topic.
Welcome!
What I'd really like to do - for now - is to take the hint, which is
currently assigned to the specific Zap channel, and somehow have it
indicate that
On 2/1/07, Andy Davidson [EMAIL PROTECTED] wrote:
What I would expect to happen, is that Asterisk would transcode
between the ulaw/alaw party, and me, wanting to listen via g729. Is
this what *should* happen ? Worth noting that my provider does not
support G.729. Is what is happening a bug
On 1/30/07, Benko [EMAIL PROTECTED] wrote:
Hello!
I've upgraded from 1.2.9 to 1.2.14 recently but experience an
unexpected behaviour with musiconhold: While in 1.2.9 musiconhold was
playing continuous on sequential extensions after a
timeout, it is restarted for every extension in 1.2.14:
On 1/23/07, Ed W [EMAIL PROTECTED] wrote:
I appreciate your point, but it's not that hard to avoid having the 9
prefix at all (in a simple dialplan at least). So to be honest one
might as well dump the whole dial 9 thing completely in the scenario
you describe?
I originally setup without
On 1/12/07, Pierre du Plessis [EMAIL PROTECTED] wrote:
Thanks Eric, I'm using the asterisk DND
Is this really Asterisk, or is it Trixbox/FreePBX/[EMAIL PROTECTED]/etc?
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On 1/17/07, Victor Perez [EMAIL PROTECTED] wrote:
Tried that, it didn't work but maybe I didn't configure it right. Anyways
how can I route all outgoing calls from that specific extension to use that
trunk?
Put that extension in a different context.
On 1/12/07, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
I am having a weird problem with one of my incoming lines. After a
reboot everything is fine if I disconnect the line from the wall and
reconnect it. After an hour or so the lies goes busy but no indication
of this shows up on the Flash
On 1/12/07, Chuck Bunn [EMAIL PROTECTED] wrote:
I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone
to the line we get a busy signal...
There was something similar to this posted a few months ago. What country
is this in? I believe the similar problem was in the UK.
On 1/9/07, Dovid B [EMAIL PROTECTED] wrote:
Hi List,
I am using asterisk 1.2.14 with real time and I am trying to send the
email to more than one email address. In that field I put in
Send the email to an alias on the system and then have the alias point to
the two email addresses.
This
On 1/2/07, Bill Gibbs [EMAIL PROTECTED] wrote:
Echo cancel: yes (and zap show channel confirms it's enabled)
I would think if echo cancel was the problem incoming faxes would fail as
well?
This is only a guess. The Sangoma is detecting the fax when it receives it,
and is turning off echo
Change step 2 on your internal extensions to do whatever you want to do
(change the ringer, callID, whatever) then go to main-aa,s,1. Or, change
step 2 to go someplace else, at somplace else, do whatever you want to do,
and then go to main-aa,s,1. The second method is easier to change if, later
On 12/18/06, Anthony Kava [EMAIL PROTECTED] wrote:
Greetings,
Back in September someone asked about documentation for the new SLA
feature
in 1.4, however they received no replies. I thought I might ask the same
question now in December. Apart from sla.conf.sample and a few comments
in
On 12/17/06, Michiel van Baak [EMAIL PROTECTED] wrote:
You can also use the devicestate commands in BRIstuffed asterisk.
That's what I use.
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On 12/13/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Does anyone have the pickup application working? I'm attempting to get
I did have it working.
The problem I'm having is in the fact that my phones register with mac
addresses instead of extensions, so I'm unsure as to what to put in the
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming
On 12/6/06, John Novack [EMAIL PROTECTED] wrote:
Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run
into some gotcha down the road where there is some missing file that
needs to be put who knows where.
Wow! Are you sure about that?
On 12/6/06, Paul [EMAIL PROTECTED] wrote:
Time Bandit wrote:
The TV ads promote it as unlimited. If there are real cases where
residential subscribers did not get unlimited residential service for
the advertised price, why aren't any state attorney generals going after
vonage?
Vonage
On 12/7/06, Paul [EMAIL PROTECTED] wrote:
Some things are clear and some things not so clear. I couldn't find
anything where specific limits on minutes in or out are stated. I think
they try to limit the number of accounts cancelled strictly for high
minutes. Accumulate enough of those and a
On 11/29/06, Paul A Brown [EMAIL PROTECTED] wrote:
Hi Mattias,
That is what I did for my 7960 and what I need to do for this. However my
problem is when I un tar the cisco file it won't run. I think it needs call
manager :-(
You apparently downloaded the wrong version. I don't know what
Didn't Digium and Polycom recently announce that Polycom phones are the
official phones for Asterisk or something like that? If so, can we at
least get the full functionality of Polycom's phones in Asterisk. Unless
I'm mistaken, the PARK soft key doesn't work with Asterisk, neither do any
of
Either write what you want, or learn to use what we have and hope
that SLA when it appears is better. Parking is not the best solution,
I think that's the problem with the Asterisk community right now. Anytime
something is suggested, the response is either write it yourself or deal
with
On 11/29/06, Brian Capouch [EMAIL PROTECTED] wrote:
Complaints are always considered, but calling the developers childish
and repeating that complaint over and over in an email isn't likely to
do much to advance the cause you've taken on.
Sorry about the rant. I apologize for making the
I am running on CentOS 4.4, Asterisk 1.2.10, hylafax-4.3.0-2,
iaxmodem-0.1.10.
I'd definitely upgrade to iaxmodem-0.1.14 and try that. The only time I've
noticed the everyone is busy is when a channel is actually busy. I'm only
running 7 channels on my setup. It looks like I am using
Can anyone point me in the direction of a
WAV or ULAW recording of those names?
http://www.digium.com/en/products/voice/allisonsmith/
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I've noticed sla.conf in Asterisk 1.4. I'd love to test it, but how does it work? There's bupkiss docs, and until I have a clue how to use it, I can't test it.
Did you ever find out anything on this? All I hear is people wanting us to test and test. How the heck do we test when we have no idea
I'm still waiting on the 2.0 firmware from Voipsupply. No luck. Don't hold your breath, I would have died a couple of weeks ago.On 11/4/06, Eric Bishop
[EMAIL PROTECTED] wrote:I second that request.
On 11/4/06, Kevin Bockman [EMAIL PROTECTED]
wrote:
Hi,Would anyone be kind enough to send
Besides ranch networks and borderware, what other SIP aware firewalls
for the SOHO/medium market exists?
Anything Cisco
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You can put the Asterisk system in front (i.e., betweenthe PSTNand your Comdial system). This will let Asterisk choose whether the call should go out over the PSTN or the Internet using VoIP.
You would use the same for the second location, provided that is a complete Comdial system. You could
On 10/29/06, David [EMAIL PROTECTED] wrote:
I looked. There's nothing there.I even did a search under /etc/asterisk for files containing Asterisk PBX and New VM (both part of the Pager message) and it didn't help. I suspect that it may be in the code.
I would suggest looking again. If it
Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My
manager.conf has an entry.
Check to make sure the file is
So, What´s your recommendation for a production environment? I waslooking for good prices, good voice quality for USA Origination and I´d
like to hear about good experiences
PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry.
I'm still wondering how this relates to the asterisk-users list. Take it elsewhere. Just like on IRC, take it elsewhere. Don't waste my time.
On 10/23/06, TV Guy [EMAIL PROTECTED] wrote:
For the record:The Digium people follower's think their shit doesn't stink. I wouldreally like to see their VC
This might be a newbie question...
You're right, part ofit is. I don't mean to sound rude, but you really need to go do some research first to get the basics down. First place is to read the book, Asterisk: The Future of Telephony (available for free, there's this site called
google.com that
If 's' is the correct extension, as Iexpect it is, how do I get the DID number that the call came in on?
What does ${DNID} give you?
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Have you looked at http://www.didww.com/support/index.php?_m=knowledgebase_a=viewarticlekbarticleid=3nav=0,1
yet?
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So I was wondering is there a way to make this happen in asterisk??
Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did
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