On Wed, Jun 24, 2015 at 2:51 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
On Wednesday 24 Jun 2015, Ivan Demkovitch wrote:
Hello team!
I’m planning to add fax functionality to my PBX. From research it seems
that there is 2 options: spandsp and Digium. I lean towards Digium app,
On Mon, Jun 21, 2010 at 7:32 AM, Ryan Wagoner rswago...@gmail.com wrote:
I saw the following lines in the log this morning. From my router logs
I see that the connection went down as my ISP was doing maintenance
for a few minutes last night. I can understand the external
registrations timing
On Tue, Mar 25, 2008 at 11:19 AM, Jiffy Slides Leonard Burton
[EMAIL PROTECTED] wrote:
HI,
We need to get our number into the White Pages.
Has anyone here actually tried it?
It's not just Voip numbers. We've got a PRI from XO that (even though
they say otherwise) we can't white pages
On Tue, Mar 25, 2008 at 1:39 PM, Andrew Kohlsmith (lists)
[EMAIL PROTECTED] wrote:
On March 25, 2008 02:15:42 pm Lacy Moore wrote:
I think that is one of the biggest things that businesses overlook
when switching to Voip. It's hard to get in the directories.
I have to say that it's been
Just chiming in here...
There was discussion a LONG time ago about a setup similar to this. I
can't remember the details, since this was something I didn't expect
to ever run into myself (I guess I'm self-centered on that one).
Anyway, the discussion was along the lines of setting the line up as
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On Dec 5, 2007 8:09 PM, Alex Balashov [EMAIL PROTECTED] wrote:
On Wed, 5 Dec 2007, Lacy Moore wrote:
the one you are logged into. Same as Asterisk. I can carry a phone
with me, and plug it in and access my Asterisk server. I can login
using softphones. Whatever phone I am on will ring
.
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and there of spare
time :-)
thanks
mark
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.
I've looked at the sample config files for 1.4 and nothing seems to jump out
at me as to what the problem could be.
For the purposes of figuring this out, I'm using Zaptel 1.4.6 for both 1.2and
1.4.
Any clues?
Thanks!
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On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote:
I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far
1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or
parked calls. It does work when putting the call on hold. If I revert back
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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
no
of no attorneys worth even asking the time. THe one's I know would give you
five different answers as to what time it is just to cover their butts.
To be on the safe side, we kill a tree everytime a fax comes in. I'm sure
that's patented too, though.
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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,
to dial your system, enter the extension, then
enter a code. By doing this, you can make sure that only you can open and
close the system.
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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,
?
Regards,
Paul
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] 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
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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
not seen a phone capable of this.
This, along with the one touch parking and XML capabilities, is
looking like the Aastra may be what I'm looking for in a receptionist
phone.
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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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something in the source
about it, but again, since it has been recompiled, this should not have
changed. Is there a config file somewhere that I'm too blind to find?
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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
Lee Howard wrote:
Yes, I do suspect that Digium sees things this way.
Maybe I'm too much of a free-thinker - too believing in the open-source
philosophy, but I would like to think that this is not neccesarily
true. I would like to think that they could host and support a
non-disclaimed
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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C F wrote:
Since Monday I didn't see much traffic.
gmail is having some sort of problem. I haven't gotten hardly any
messages from any of the digium lists in my gmail account.
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Jim Duda wrote:
I've been on the shorewall firewall and confirmed that I have the
firewall configured properly for VOIP QOS.
What exactly have you done here? You do know that you are apparently
using IAX2 and not SIP. Those are not the same protocols. In fact, if
you configured the
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
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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
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