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 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
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 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 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 9/7/07, 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 9/13/07, 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 9/17/07, 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 9/17/07, 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
] 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
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
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
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 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 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 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 7/6/07, 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 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, 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 1/23/07, 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 1/30/07, 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 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 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 2/8/07, 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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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 2/14/07, George Wise [EMAIL PROTECTED] wrote:
Does anyone know of a good Asterisk/LAN/PC support company in Houston, TX?
Yep
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On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
Does Trixbox support
www.trixbox.org
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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 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 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 3/6/07, 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
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 3/10/07, 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
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 3/19/07, 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 3/21/07, 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 3/27/07, 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 3/28/07, 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 3/29/07, 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 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 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
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 4/14/07, 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
Load information is the file that ends in .loads. Just use the name of the file minus the .loads.
I just checked mine, and it is the same as what you have. Do you have the firmware on your tftp server? Just so you are aware, the message indicator light does not work on that SIP image. And, if
Please search the wiki first. Most of your questions you post can easily be found by doing a search. Put some effort into finding the answers to your questions first and on your own, and then if you still have questions, I'm sure everyone would be more than willing to help.
On 8/29/06, Crazy Boy
Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this.
On
I may have to do something like that to be able to setup some way to temporarily close our office. I haven't really found anything else that would have a visual indicator that the system is on temp. closed mode. I can manually set a database entry (which I already do), and I know I can add an
Aaron, was the MWI working for you on 8.0.2? I've got a 7970 and 7961 sitting on a shelf because the MWI doesn't work. On the 8.0.4, it never registered, but I was able to make calls with it. I didn't try calling it, since I never saw it register. It appeared it was authenticating for outgoing
Not sure where you got your SIP image, but my SIP files have that particular file in it.
On 9/1/06, Jason Lixfeld [EMAIL PROTECTED] wrote:
On 1-Sep-06, at 3:41 AM, Tomislav Parčina wrote: In article
[EMAIL PROTECTED], jason +lists.asterisk@lixfeld.ca says... I've been having some problems with a
exten = 100,1,Dial(OP1OP2OP3EX1)
where 100 is the extension the caller has dialed, OP1 is the first operator, OP2 is the second, OP3 is the third, and EX1 is the executive.
on operator's phone, you could put a second line called (on the phone display) COVER or 100 or whatever. I do something
You also have to make sure that on the web config for Grandstream that you allow it to receive auto-answer (or something to that effect).
Ok, actually it's under the settings for the Lines and is called: Allow Auto Answer by Call-Info:
Make sure Yes is selected here.
You can use what Barry has
I second this wish.
On 9/14/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message- From: Norris, Sam [mailto:[EMAIL PROTECTED]
] Sent: Thursday, September 14, 2006 8:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] BLF across asterisk trunks
2 asterisk boxes
Do some 7960s perform differently?
On 9/15/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Rich Adamson wrote: Julian Lyndon-Smith wrote: I've got a cisco 7960, with (amongst many others) the following in the
RINGLIST.DAT file Foghorn foghorn.raw I can manually select this for the ringtone.
You'll need to change your XP box to the default router of the phone (or, just change your XP to a static on your local net, then add a static ipunder advanced for the default route of the phone, that way you still have access to the internet from your XP box), and then add (under advanced) the IP
I dial using groups. Dial(Zap/g1/1234)
I'm pretty sure this was taken off of the examples on the Sangoma website.
On 9/19/06, Mario [EMAIL PROTECTED] wrote:
I have a Sangoma PRI card configured for E1 line (i.e. 30+1 channels perport) and I'm not quite sure on how the Dial command should
See here: http://linux.thorsten-knabe.de/asterisk/pickup.jsp
On 9/19/06, Miloš Kocbek [EMAIL PROTECTED] wrote:
The problem is that i want to be able steal a channel without any action at extension 100. I want to be able to dial a number and talk to whoever was exten 100 talking toI hope you
It does matter if they are using Asterisk. I think we'd all like to know how they are doing it.
When I was using the chan_sccp driver, I was able to display that information on the screen. I'm not sure the SIP protocol in Asterisk supports this (and don't know if the SIP protocol itself supports
I couldn't get the hinting to work. Went back to 1.6.7, same config, and it works. I wasn't sure if the config had changed between the two. But, now that you mention it, I did experience a phone rebooting several times. I was half-way paying attention, so I just thought I had done something.
On
On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote: [9580] type=peer auth=000413242fff:[EMAIL PROTECTED]
It would be
[MAC ADDRESS]
type=peer
...etc..
Or at least, that's how I interpreted what Eric said. I think that's an excellent approach. THe phones are devices. An extension calls one or more
Has anyone implemented something such as this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+voicemail+live
It looks like it would be a good addition for some situations. I have a user that would benefit greatly. Unfortunately, there are some items that need to be worked on, and so
FYI, this is an asterisk mailing list, NOT trixbox. Most people on here don't care when trixbox is going to do something. Try their list.
On 9/25/06, Christopher Corn [EMAIL PROTECTED] wrote:
I know asterisk 1.4 has t.38 pass through, but I don't think trixbox does. i run trixbox. looks like for
Most people here don't give a damn about anything but their own personalproblems. Sounds like you are probably one of them,
Thanks for noticing!
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Be sure that it is looking in the right place. If it is running as non root, then the ctl file would be in a different directory.
It looks as though Trixbox does run as non-root. The ctl is actually/var/run/asterisk/asterisk.ctl.
Did you install from scratch, or was a previous version of
Wherever you have your exten = s,1,Answer statement, replace with:
exten = s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer
exten = s,n,Answer
then continue on.
Asterisk will then wait 30 seconds before it answers the phone. You would probably want this
Wyatt: Try dialing *97
unplug: Are you referring to the greeting that says welcome to asterisk mail or something similar to that? If so, that file is called vm-login and is in your sounds directory.
On 9/25/06, unplug [EMAIL PROTECTED] wrote:
Thanks.Are you talking about the customization of
Try vm-intro.
On 9/25/06, unplug [EMAIL PROTECTED] wrote:
In the function of voicemail, the default greeting is:Please leave your message after the tone. When done, hang up, or press
the pound key.It is what I want to replace.On 9/26/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: Wyatt:Try
Is it possible to manually set the hint status of a virtual extension via the dialplan?I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn on a light for night mode.
See
Just wondering if anyone has had any luck getting the cisco 7935 workingwith asterisk and if so, what is the best way to go about it?on the
The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP
;exten = 799,hint,DS/mmgc
Lacy, What is the DS/mmgc?
The DS is what the DevState patch adds. I actually got to this point by following this thread:
http://forums.digium.com/viewtopic.php?t=891highlight=shared+line
After I implemented these changes, I had the DevState on the system. The mmgc
Steven,
If youare trying to do this on a stock Asterisk system (and I can certainly understand why you would want to), then what I have implemented will definitely not work. I couldn't find anyway to do this on a stock system. Upgrades are going to be a nightmare with all the patches that have
Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs).
On 9/26/06, Shawn Kelley [EMAIL
I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint.
Ex:
exten = 132,hint,SIP/DEVASIP/DEVBSIP/DEVCSIP/DEVD
if I add SIP/DEVF, DEVF is not monitored.
Is anyone else monitoring more than 4 devices, and if so, what
I didn't see it as making fun of anyone. I, for one, was curious about it. I suspected it was some type of translation issue, whether it was a word in another language that doesn't translate or what. I know there are many concepts in English and in other languages that just doesn't translate
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