I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was suprisingly easy to
Is it possible in any way to use an Outlook contacts database as the
source for the internal Asterisk database that is used for callerid
lookups?
You could do it with some simple CDO or MAPI calls but the overhead would be
brutal and you'd never get the data in time unless you were running
Just documenting this issue and how I solved it for future reference on the
list, hope it helps someone:
I blew away my primary Asterisk install just because I felt it wasn't as
clean as it could be. I wanted to put on the latest AMP 1.0.007 (which, by
the way, totally rocks) and everything went
Just wondering, did you download AMP-1.10.007a bugfix release ? I have
installed it a few days ago and it went fine. (somewhere beginning this
week I guess)
Cheers.
Kristof.
I didn't note which one it was, just clicked the topmost link on the
download page from SourceForge. I did take a
What do you have to do to get * to see the TDM400P? It sees the PRI card
and associated channels
but I can't get the TDM400P to work - no matter what mix of channel numbers
I use ztcfg doesn't
like it.
My config with a Digium PRI card and a TDM400P, just finished yesterday
working fine:
Is this ok to sell this on Ebay when they are using open source software?
How is this different from an Asterisk integrator doing a custom config for
a customer and selling it to him?
I would think it would only be a GPL violation if he tried to pass off the
Asterisk install as his own creation,
I've been testing spandsp with mime-construct on 1.0 - stable with the
inbound fax number being routed as a DID on a PRI. While I have it working,
and it emails me a PDF fine most of the time, I've noticed some issues on
receive:
1. It's a little bitchy on long faxes from analog machines; it just
The AMP configuration didn't work so I decided to work up from the Wiki
example. Can anyone help?
This is a good config using AMP:
http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515
HTH
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My original assertion:
2. Sometimes I watch it spool the inbound file then hang up the Zap channel
and mime-construct doesn't kick in, like it's bailing out of the dialplan.
Seems to be correct. What happens on a long fax is that tiff2ps does not
execute in time for mime-construct to kick in,
: Tim Litwiller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 4:45 PM
Subject: Re: [Asterisk-Users] IAX2 - Between two ASterisk Servers
Colin Anderson wrote:
The AMP configuration didn't work so I
2. Here is my macro-faxreceive
Boris, thanks for the great reply. I was thinking about this all evening,
and just finished off a far more modest script that does the trick. I just
sent 10 simultaneous inbound 100 page faxes and they all went through fine.
Here is my (simple) solution to increase
this in the user section of the IAX.conf. It points to the same
context the zap channels use. I am trying to dial a SIP extension.Any
ideas?
Regards,
Chris
- Original Message -
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk
Title: IAXy Provision
http://dacosta.dynip.com/asterisk/
enjoy
-Original Message-From: Wiley Siler
[mailto:[EMAIL PROTECTED]Sent: Wednesday, April 13, 2005
10:02 AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] IAXy
Provision
Hello
In /etc/crontab:
0 2 * * * reboot
Runs the reboot command every morning at 2 AM. Works for me.
If you are using a RedHat / Fedora box and you have cron running at startup
as a service your crontab entries should be running as root, so it should
work fine.
cron is super bitchy about
On Tue Jan 04, 2005 at 01:27:21PM -0500, Philippe Daoust wrote:
Anybody know anything about this F-1000 phone?
100 hours of battery life, not bad at all...
The peanut gallery chimed in on this yesterday:
http://slashdot.org/article.pl?sid=05/01/04/1816228tid=193tid=215
http://it.slashdot.org/it/05/01/12/1829240.shtml?tid=215tid=218
http://blogs.zdnet.com/Ou/index.php?p=25
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Assumptions:
-Working DHCP server
-Good LAN, everyone's happy and can see each other
-Working DNS server on LAN
-IP of Asterisk server: 192.168.1.46
sip.conf: (remove comments)
[550] 'extension number
callerid=Joe Blow 550
canreinvite=no
context=from-internal 'default AMP context, salt to
I called Telus before Christmas requesting some sort of VOIP connection.
We are going with babytel. I'll advise how that works when it is up and
running, hopefully next week.
[plug] www.thinktel.ca
I know the guys they are competent they will sell IAX. Peered thru GT in
Downtown Edmonton.
Ouch ... error while writing audio data: : Broken pipe
Did you run make samples from /usr/src/asterisk?
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Why do people want to develop more softphones? There's already lots of
softphone projects out there that could use a hand, if you want to work on
one. Starting from scratch will just add one more half baked softphone to
the growing list of unuseful Open Source applications.
Especially since you
The hack came in through ssh.
IMO, your best defence is an extremely strong root password; I am often
mortified by looking at my logs and seeing all of the login attempts through
SSH.
OT: I am not up on Linux script-kiddie type tools, but I assume that there
is a script of some sort that
Thanks, everyone, for the excellent suggestions.
For posterity and for future reference when this thread comes up again,
summarizing the best way(s) to defend against SSH logon attempts:
1. Don't allow root thru SSH or Telnet, force logon as regular user and sudo
2. If you must run SSH or
I've heard comments about the instability of the IAXy (loosing IP, locking
up, etc) . I want to deploy some, in a remote site. The comments on the
instability suggest that the simplest approach is to power-cycle them when
you have problems. The IAXy would only be in use from 3PM to 8 PM. My
I hesitated before sending this, as I have been flamed before for being a
beginner. but
I am newish to linux/asterisk, and I am running an ssh server. It is still
running with default settings, (I dont know yet how/where to change it),
and I CAN logon remotely as root.
Debian: Yes, root login by
Why would someone choose these over other
boxes, such as the Sipura 2000 and 3000?
Because I want
NAT traversal and a low bandwidth codec. That's the whole point of IAX2 as
opposed to SIP.
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Has anyone had any experience with wireless LANs and Asterisk?
I have played with the LocustWorld distro but not at length. Basically, it
works. Some sort of QoS tagging for SIP, the docs on it are scanty. It has
it's own internal encryption. Never tried it in full force, mostly because
of
John Novack wrote:
And the only IAX2 box made is the Digium one, with it's current
shortcomings ?
From reading through the archives, it seems there is currently no way to
reset to factory default, no written MAC address on an individual box, and
some other instabilities requiring frequent
Extension 's'? I thought 's' meant Start, not an actual extension. If
there's something I'm not reading or need to read again, don't
hesitate to hit me with a clue stick.
Sort of. 's' is used when there is no matching extension in the context.
It's the fallback extension if there's no match.
http://advancedippipeline.com/60400413
BOULDER, Colo. -- Leading Voice over IP service provider Vonage Holdings
has complained to the Federal Communications Commission that competitors are
blocking the use of its service, according to FCC chairman Michael Powell
and others close to the company.
Yeah, I'd like to hear you guys' opinion instead of CleverNickName's!
-Original Message-
From: Luki [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 15, 2005 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port
how does the phone know where to find the TFTP server..?
Dude, option 150 in your DHCP server:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a00800942f4.shtml
We use the same option for our Mitel phones. HTH.
___
I get sideband (interpreted by the enduser as echo) when calling a POTS line
through our PRI on Snom's or X-Ten. Our crappy BudgeTones actually sound the
best.
Everyone is exactly right here, that it really depends how the call is
terminated on the other side. Calling from a Snom terminating to
Dean,
try the command "find / -name dialparties.agi". The slash means start searching
from the root directory.
On my
FC2 box I get:
/usr/src/AMP/amp_conf/var/lib/asterisk/agi-bin/dialparties.agi
and
/var/lib/asterisk/agi-bin/dialparties.agi
hth
-Original Message-From: dean
I googled on this for about an hour and the most relevant hit I got was, of
course, the first hit:
http://www.sowerbutts.com/linux-mac-mini/#support
In it, he indicates that the stock Bluetooth module should work, but
untested - he doesn't qualify the statement with anything. Has anyone tried
Any assistance would be greatly appreciated.
If your install doesn't have to be 24/7, cron a reboot at 2 am. Since I did,
it's a nonissue for me.
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We have Bluetooth working up to a 75 foot radius from the base station
through standard drywall and plenums. Dunno if it is localized to those
specific devices, or whatever. I am specifically looking at this to issue a
single handset to our sales people who are in and out of our office, and go
to
I use bluetooth for presence detection in exactly this manner. Sadly, I'm
stuck in Linux because I also use a TDM400P in the asterisk server. I
think
the Mac Mini would be a great solution for this sort of feature.
I promise to report back to the list and document on the Wiki my
experiences, if
This wiki should cover most of the basic stuff that gets asked over
and
over again just to help reduce the amount of repetition that most of you
have probably noticed takes place here.
Problem is, Wikis in general suck and voip-info.org in particular is quite
useless except as a random
Lets point them to google site:voip-info.org
or site:lists.digium.com then. We do a lot of that once they get on the
list. Why not before?
OK, Ollie J if you are listening maybe you might consider appending those
links to your monthly or weekly list etiquitte reminders. Post them daily,
even.
*spews coffee over keyboard*
- FUNNIEST - THREAD - EVER -
Also one of the most insightful.
Teddy, your gmail invite is on the way.
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This list is for discussions among users of Asterisk, not a getting
started hotline for beginners. Beginners learn by reading documentation
and examining the sample files included.
Mmm, I (respectfully) disagree. One of the unstated objectives of mechanisms
like this list is to evangelize the
2 parts, 1 to Andrew, 1 to Mr Critchfield:
Andrew's part follows:
If I am having a particularly bad day I might
jump the re-offender part but that is *very* infrequent and I often
apologize
publically afterward. We are, after all, all human.
First of all, thanks for the well-reasoned
What is the best Asterisk manager to use, i do not mind web based or GUI.
AMP
http://amp.voxbox.ca
Although the best Asterisk GUI is only a fraction of the power of the whole
thing. If you want a little, regular PBX, AMP is just the trick.
If you want to do anything fancy, all of the GUI's
Can anyone tell me why these fail each time?
I can, but I won't since this is a FAQ - google it!
Also what is the latest revision?
goto www.grandstream.com
See, this is exactly what I'm talking about. How churlish. Why did he even
reply? I have a BT100 sitting on my desk, and it took me all
The CTO is highly unlikely to know or care about the low level technology
decisions. It isn't something that bubbles up to his level or pay
grade.
Unless it's presented to him as a means of doing something faster, better,
more cheaply, better interop etc. Isn't that the value proposition in
Take a look at eqso.org
Aside from FRS, interesting thing about this, is that they are very careful
to treat the VoIP part under the same rules as FCC HAM i.e. if I were in
Canada and I was using, say, Asterisk or VoIP in general to hook into this
thing with it's transmitter in the US,
Pure guess... in the US, probably treated like rtty?
Interesting thought. There's no 'r' part in the 'tty' part in this case,
though (unless you were transmitting rtty through VoIP).
This whole thread opens up all sorts of interesting ideas. Chris A's musings
are interesting as hell too. My
I'd like to find a way to have my asterisk server in a DMZ protected
from outside and not directly on the internal network. Is there any
recommended architecture ?
One of my current installs is a DMZ with an * server protected from outside
and inside with Monowall:
http://www.m0n0.ch/wall/
I'm trying to formulate a strategy for our interconnected Asterisk IAX peers
to failover to the PSTN in the event of a DDoS. We currently use them like
this:
DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP
This works fine, and everyone is happy. One of my concerns, though, is if we
How about a combination of GotoIF, and app_dbodbc (or app_db):
exten = 700,1,playback(ddos-on)
exten = 700,2,DBput(DDOS/yes)
exten = 701,1,playback(ddos-off)
exten = 701,2,DBdel(DDOS/yes)
[mymainaa]
exten = s,1,DBGET(TRUE=DDOS/yes)
exten = s,2,Do this
exten =) s,102,do something else
My
Are these inbound or outbound calls? (both?) I am pretty confused
about all of this...
Sorry, I should have been more specific. The primary Asterisk box that
connects with the PRI is the one I am concerned about being DoS'd - the
remote IAX peer runs off of a cable modem with a dynamic IP, I
I can not even get IP anymore from my DHCP
Hate to ask the obvious, but is the DHCP server on the same VLAN?
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] Grandstream and VLANs
Yes :)
It's not DHCP as the phone won't work even with statically assigned IP. It
basically looks like Grandstream is tagging and/or reading the tagged
packets incorectly.
W
- Original Message -
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non
to untagged vlan and
leave the default VLAN 0 on the phone then everything works just fine
W
- Original Message -
From: Colin Anderson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 4:38 PM
Has anyone had production experience using * w/ MySQL Blobs to store sound
files? The application I am working on requires all user data resides in a
database. I am currently reading/writing the files to disk via a phpagi
scripts but I would love to read the blob into a variable in the dial
I've read through a good amount of documentation on voip-info.org, but
hadn't found a solution, so I thought this list might help. I'm not
great with linux, and I suspect there might be a port problem... maybe
Asterisk isn't listening for SIP clients. How would I go about
checking this? X-Lite
FWIW:
I order a lot of Dells. My boss is cheap. That being said, I *like* Dell,
it's a very well designed box. It's been said many times that Dell does not
innovate, instead they copy and improve and I firmly agree with the
improve part - they are a dream to work on.
Some things to watch out
Title: Channel Banks
We use
an Adtran Atlas 500 for this job (not for * but for our Mitel ICP 3300) you can
aggregate FXO to T1 / PRI or any which way you want. It's a killer box and very
easy to work with. Adtran support is, in a word, phenomenal.Very pricey,
but ebay has some 800 models:
That'd be the 3CNJP24SE, we have one that powers 3COM NJ-200's. Works well.
-Original Message-
From: Martin [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 9:23 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Power Over Ethernet for *any* ethernet
switch (or hub);
If you look at the specs on the Dlink box that Primus gives you, you will
see that it is SIP.
I am sure Primus has a SIP platform because we have played with it. We
managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2
hard phones. Their PC-Phone app is also a SIP soft phone.
Probably be easiest to use something like an Eicon Diva T/A which breaks
down a BRI to 2 pots or 1 data and 1 pots or a 128K data- you'd use the Diva
as a bridge to the BRI. It'd sorta look like:
PSTN
| |64K serial-Your PC
|Diva |
|
I like the way the 3com NBX system works. The web interface is pretty
intuitive. Adding users and devices is a snap through the GUI but to get
to
the real meat you have to edit the dial plan. To do this, you download a
text file to your desktop, edit it, then upload it again.
Ditto on the
DID's from Allstream (ATT) are $2 Cdn/month but I think they have a rule
that it has to terminate on their network somewhere...
-Original Message-
From: Linus Surguy [mailto:[EMAIL PROTECTED]
Sent: Monday, June 14, 2004 6:53 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Canadian DID
Second that. Using stacked HP 2650 switches to support ~120 users and with
QoS on, don't even have to worry about VLAN'ing the network. In HP/Compaq We
Trust...
I have been quite happy with our HP 2848 GigE switches that we put in
for our desktops a few months ago. I have also used the 2650 48
I am also getting a call not approved error on xlite??I know a fw
people have also come across this problem because Ive seen threads
posted on it but the solution has never been posted. If anyone has
idea please let me know.
We ran into this problem too and we found that the settings for Asterisk
Hi,
thanks for responding. For the record, I did get my 2 M20's running 1.0 stable
with a T100P and a TDM400P with Fedora Core 2. Installation was straightforward,
and even facilitated by IBM's great design - there was no problem allocating a
power plug to plug into the TDM400P for example,
Short
answer: No.
Long
answer: There are GUI's that allows for *most* basic configurations but they are
usually incomplete in their implementation so at some point or another, you
*will* have to edit a .conf file.
The
closest I have seen, and the one I use for day-to-day is AMP at
Just wondering how difficult it would be for AMP devs to develop a
install wizard or a batch file that can automatically execute the
install and download necessary dependencies... until then, I guess
I'll be continuing to manually config my asteisk files
IMO the brutal part was
We use several Dell 2650 servers. Order
them with the dual DC power supply option.
Buy a row of -48 batteries and a
-48 power source, your servers will stay up for hours.
That's only half of the solution. How will the phones be
powered? Some thoughts:
-If your power is iffy
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
opinions:
1. asterisk -p == renice -20 ??
2. I've turned off swap with no apparent ill effects. Can anyone commment on
long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day)
3. Can anyone comment on
== Spawn extension (default, 998004, 3) exited
non-zero on 'SIP/2004-41dc'
What is the meaning of the exited non-zero line?
afaik, after something executes, zero is returned: Everything went OK or
-1 Something bad happened, you can branch conditionally in the dialplan
based on that.
I have exactly this problem. When it happens, I lose access to some FXS
ports and get Geiger counter style clicking on the FXOs. I just opened a
ticket with Digium on the subject, but given what I just read, perhaps I
should not have high hopes.
Kinda same with me; although on some of my FXS
I believe that the cause of the error
is related to the line, Ring requested on unconfigured channel 0/23
span 1.
Dec 6 04:19:43 WARNING[4891]: Ring requested on unconfigured channel
0/23 span 1
I don't have a silver bullet here, but isn't it wierd that it's indicating
Channel 0? On my PRI,
If five people in the office all need to use their phones at the same
time, would I need five VoIP lines, or would I only need one VoIP line?
Am I over-thinking this?
You would need 1 broadband connection, and technically, you would need only
1 ACCOUNT (I think that's the word you are looking
EXACTLY ;)
FWIW, one of the things I was was trying to illustrate was the difference
between per-minute and unlimited. In the original poster's scenario, I think
what he was driving at was having 1 user using an unlimited account among 5
guys, thus cheating (negative word: try maximizing) a
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User (0)
Ext: 1 Cause: Info. element nonexist or not implemented
(99), class =
Looks like the switch on the telco end might be confused about the number
pattern; try adding:
pridialplan=unknown
to
I don't really understand why do many people set their pridialplan. The
text PRI Dialplan: Only RARELY used for PRI. in zapata.conf.sample is
not there to just take up space in the file.
Some telco's do, some telco's don't. The text Rarely used for PRI is IMO a
location-centric and
Is there something different that I should be dialing?
On my pri, I have dialplan=national. It is from Allstream/ATT co-lo with
Telus which pretty much behaves the same as most US carriers. We don't dial
international, so it's not an issue for us.
If I dial XXX-XXX- with local area code,
Also, the box is on a UPS, so I'm assuming the AC power is generated
from the battery and is a perfect sine wave. But that begs the
question: does a UPS system connect the mains to the output, or is the
input power used to charge the battery, and the battery used to generate
the output
Any thoughts/suggestions would be greatly appreciated.
Adds, moves, and changes which are the bane of any telephone administrator.
Show how fast it is to add an extension with voicemail. Using AMP, I can add
a new SNOM in under a minute with voicemail. Contrast that with the 15-20
minutes or so
2 weeks ago,
thanks Peter S, Gilad Ben-Yossef, Joe Greco, and yes, even Mr. Critchfield:
On Thu, 25 Nov 2004, Colin Anderson wrote:
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
opinions:
1. asterisk -p == renice -20 ??
[Peter S]
The -p option sets asterisk
I did that already but it did not work. Thanks
The issue might be the FQDN of the * server itself. If you are using a
smarthost under your control for relay, the host name should be substituted
automatically with the smarthost's FQDN. If not, then in your
/etc/mail/sendmail.cf:
DSmachine name
Thanks for the reply.
This * box currently sits on a WINDOWS network behind a firewall and
does not have a FQDN.
Still should work. Try it. Sendmail attempts to resolve it's hostname
against dns and it's own hosts entries, and if it can't, it puts in whatever
it can, even localhost. The DS
I'm thinking in sending a mail for asking WHY THE HELL they can't support
bare modems, even if they have voice support
Smart move, on their part:
1. Digium exists to sell hardware. Without hardware sales, formal Asterisk
development would stall, and the project as we know it would fragment.
Check with your telco. We had the same problem on 1 of our PRI's, every day
at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly.
Turns out the equipment at the CO was going into a test loop at that time
because of a forgotten setting by a tech. Man, what a finger pointing
Should I set the channel bank to
provide timing or receive timing?
My Atlas 550 provides
timing and I set my zaptel.conf to:
span=1,0,0,esf,b8zs
Never had a red alarm, ever.
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You have either miss typed or have been lucky.
The former is true. Should have been 1. My apologies. Posted as plaintext
per your requirement.
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Anything I should know or do
that would make this go easy?
Yes:
http://www.xorcom.com/rapid/
this distro rocks.
Not Gentoo but was totally brain dead easy to install. Did it yesterday, and
had a couple of softphones working in *45 minutes* starting from configuring
the RAID array to dialing
-Using FC2
-Digium T100P card
-lsmod | grep wcfxo yields:
wcfxo 16288 0
zaptel 232580 3 wct1xxp,wcfxo,wcfxs
so my drivers are cool
-zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
I've got some DID's on a PRI that I want to ring an extension when the
trailing digits match an extension in the dialplan.
Followed the instructions at:
http://www.voip-info.org/wiki-Asterisk+tips+DID
When I call the DID, Asterisk is not recognizing it as a DID and instead
sending it to the s,
put an underscore and see if it helps:
exten = _0545,1,answer
Thanks for replying.
Good idea, but no effect.
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can you post what you have regarding these channels in zapata.conf?
Thanks for replying:
[channels]
context = from-pstn
switchtype = national
signalling = pri_cpe --the Adtran box looks like the network, so we set it
up as cpe
group = 0
channel = 1-23
immediate=no
overlapdial=yes
If I set the User Term on the Adtran to only send 7 digits, the PRI debug
says:
Called Number (len= 7) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0545' ]
So, looks like the DID is being recognized, but still the same problem with
hangup when the
Fixed my own problem, for the benefit of the list archives: Recompile,
reinstall.
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Lol, in other words for newbies forget it, I've tried like 3 or 4 times
to install it and their documentation just plain sucks, so I'm prepared
to pay anyone $200 that can write a step by step guide and then this can
be posted online to help out others.
Hey, I'm a newbie and I got it to go
On November 12, 2004 09:46 am, Rich Adamson wrote:
As far as the motherboard issue, its not just the digium products that
have an issue. If you know of someone that is heavy into using audio
applications (eg, song writers, midi stuff), they have known about the
pci / interrupt latency issues
There is a reason Microsoft is winning guys and price isn't it.
At the risk of starting a flame war, I agree, although it's not nessisarily
a bad thing. There's two fundamental reasons why documentation and the user
experience (UI inconsitiencies for example) with Linux basically sucks:
1. The
So, if anyone is interested, I am suggesting particularly a standalone,
cross-platform project that is simple to install, configure, operate and
manage. It should operate with or without a database. It can leverage
existing projects, but it must not have the existence or installation of
those
I have used this device with good results:
http://faxswitch.com/stick_fax_phone_modem.html
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unknown usually indicates that there is active communication between
Asterisk server and the phone that does not involve a voice codec. An
example of that would be registering the phone. Another example would be
call setup and teardown before RTP data is sent and codec preference is
established.
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