See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003976.
Essentially, the bug is that if a callback agent puts a caller on hold,
that caller does not hear MOH. This bug has been around for a while,
but nobody has been able to follow through on testing to the point where
we could
Jim Van Meggelen wrote:
I would like to start a discussion centred around the various ways one
might serve up configuration files from an Asterisk server (I know, it's
[snippage]
I have heard that khttpd is pretty lightweight, but its use seems to
have been deprecated, and it does not appear to
Mark Floyd wrote:
I am trying to get the name and number to show up for an incoming calls on
my Polycom IP 500. Right now only the name shows up, but in the call list
both name and number show up. Any help on what to change in the config file
would be greatly appreciated.
Watch the display.
Robert Rozman wrote:
Hi,
I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is
restarted. Can this be avoided in some way ?
As Kevin insinuated, there is support for this in CVS Head. It's called
persistentagents and is set through agents.conf:
; Define whether
[EMAIL PROTECTED] wrote:
If I remeber correctly, Mark Spencer is working on encryption in IAX2
Sort of. Some IAX encryption code went into CVS a while back, but it
was more of a talking point than anything else, meant to give
interested developers a starting point. The -dev and -security
Mike Sander wrote:
Hi
I know that attended transfers are only available in the CVS Head.
The version info reports:
Asterisk CVS-v1-0-02/03/05-10:24:22
You don't have HEAD. Follow the instructions on asterisk.org to
download from CVS HEAD (as in, not -r v1-0).
Nick
Bruce Komito wrote:
That's your opinion, and I'm sure you have good reason for it.
However, in order to be widely accepted, any app must support mysql,
simply because many environments run mysql as their choice of
database, and are not likely to change.
Ah yes, the pack mentality logic for
Andrew Kohlsmith wrote:
On January 23, 2005 04:04 pm, Mike Sander wrote:
Is the harddisk activity on a standard asterisk install such that I
don't really have to worry if the power cuts??
Not typically; there isn't much writing going on, this is true. Are
you that cash strapped that a $75 UPS
Kanwar Ranbir Sandhu wrote:
On Sat, 2005-22-01 at 12:19 -0500, Jim Van Meggelen wrote:
I'm curious to know how the volume of Asterisk-Users rates as far
mailing lists go. This list sees over 200 messages per day, which has
GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has
[EMAIL PROTECTED] wrote:
Hi Guys, Gals.
Ok, so I have latest CVS sources on a debian box, 2.6.10-1-386 kernel
kernel headers isntalled in the right plauce and all that stuff .. but
whatever I try .. same results, I only need to get ztdummy working for
a conference .. but I always end up stuffed
Greetings all-
For whatever reason of personal insanity, I've decided to start an
Asterisk bookclub. Basically, we'll pick three books every month (a
users book, a developers book, and another general interest book) and
then read and discuss on IRC in the #asterisk-bookclub channel.
The
Serge Schumacher wrote:
Hi,
Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such
switch 'Realtime'
What does this message mean ?
Something wrong with the switch statement in my extensions.conf or
maybe is the module net correctly installed ?
Perhaps you might
Alspach Family wrote:
Today is the day. The most up to date list is attached. I will
forward it to Rob Friday night so, anything you want and can get to be
by then I will add.
Do the phrases being send include those in bug 3006? If not, can they?
Lets say half of them are active readers and
Gabriel Afana wrote:
Ahhh, and I've read every message telling everybody they dont have the
lastest version...thats why I went to asterisk.org and downloaded the
highest-number version I could find in the FTP Okdownloaded latest
CVS but now asterisk wont compile. I had it working before.
Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls. This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call fails every
time with
Nick Bachmann wrote:
Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls. This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call
Tracy R Reed wrote:
On Thu, Dec 16, 2004 at 01:11:34PM +0100, Roy Sigurd Karlsbakk spake thusly:
I think there is a bit more difference. The byte code of ulaw is a
monotonic function of the amplitude whereas in alaw the code is xor:ed
with a bit mask of 0x55.
Wow! Encryption!
Scary
Satchid wrote:
Dear Members,
I am searching for a new PBX for the company. My choice is Astrisk. My Boss
wants background music via all the telephones. This is done in a
conventional PBX that he wants, but I can use the Asterisk PBX if it can do
this also.
As I said he needs background music on
Matt Riddell wrote:
Thomas Johnson wrote:
Hello-
I've got some audio CDs that I'd like to use for MOH.
I just thought I'd point this out seeing as no one else has.
It is illegal to use an artist's music for music on hold without the
copyright holders permission.
Since you brought it up, I'd
[EMAIL PROTECTED] wrote:
can you make asteirsk do a fast ring as well?
Do you mean a fast busy? If so, what you're looking for is called
congestion.
If you mean making a faster ring cadence (time between rings), look at
the indications.conf wiki page.
Nick
P.S. Please don't post that
nik martin wrote:
Anyone ever thought about an Ethernet based channel bank? Basically a
rack mount set of 24 IAXys? That would be cool, IMO. No wrangling
with zaptel, etc. IAX as the * - Channel bank protocol.
These exist. The Mediatrix 1124 is just one example.
Nick
Matt Schulte wrote:
All,
We are using a SIP provider that is expecting 0-15 response for
fmtp. Our CVS Head asterisk server is sending 0-16, I looked up an rfc
and it stated:
RTP Payloads for Telephone Signal Events
RFC 2833
Henning Schulzrinne, Scott Petrack.
May 2000
[EMAIL PROTECTED] wrote:
For an office that is using VoIP phones to connect to Asterisk, is
gigabit ethernet really necessary for the Asterisk box to connect to
the switch? I know that I won't even approach the limits of 100 Mbps,
but would gigabit help with latency / collisions when several
Noah Miller wrote:
I've told lots of people about the Flash Operator Panel before, but
I've never actually used it myself. I've got it all set up nicely,
but I can't seem to authenticate to the asterisk manager (which is
running on the same box). When I set the op_server.pl to give debug
Sean Cook wrote:
Sounds like it is time for a different router... There are a few routers
out there with buggy nat engines... they are fine when you are doing
typical nat, but if you are trying to do 1:1 nat... get a good router or
I've been very happy with Netopia 3386-ENTs.
make a BSD box to
[EMAIL PROTECTED] wrote:
*Smack*, you're right, changing the g3 to g2 help nicely.
But now the PRI seems to be refusing the call (Channel 0/1 got hangup):
--snip--
-- Executing Answer(Zap/38-1, ) in new stack
-- Accepting call from '' to '15123455476' on channel 0/14, span 2
Nov 28 16:08:14
Rich Adamson wrote:
I'd certainly agree on the IP500. Got one sitting in front of me
right now. Works great, but learning curve on configurations is
greater then the Cisco's partially because all config parameters are
in xml format
I agree, the configuration files are hard to learn, but they
Matthew Boehm wrote:
Hey guys,
Looking for some suggestions here on hardware to use. We have several
business customers wanting to start using our VoIP service. We will be
replacing their 10 year old Panisonic system with Cisco 7940 phones and a
T1. Problem is, they have to keep 1 POTS line for
Matthew Boehm wrote:
Could you give an example of a cheap 1-port FXO gateway ? And yes, the
Cisco phones do have an Emergency Proxy and a Backup Proxy option.
(Please reply inline instead of top-posting so that threading is preserved)
I believe the SPA-3000s have an FXO port. If you want a
Garry Taylor wrote:
Hi Steven,
Please don't top post, it breaks threading.
-- Zap/2-1 is ringing
Nov 27 13:25:27 WARNING[6718]: chan_zap.c:3463 zt_handle_event: Didn't
finish Caller-ID spill. Cancelling.
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
Open up chan_zap.c, change
michelle li wrote:
Hi:
I am new user of Asterisk. I can make phone calls. How
can I stop the call after it goes a certain time(eg,
20 seconds). I tried AbsoluteTimeout command, it does
not work. Can anyone help me?
Can you post relevant sections of your dialplan... I suspect you do not
have
Kevin Brennan wrote:
Kevin Brennan wrote:
I am planning to configure * box A with PSTN interface to route faxes to
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
2GB?!? Remember, each voice channel you trunk across TDMoE is
Kevin Brennan wrote:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
2GB?!? Remember, each voice channel you trunk across TDMoE is 64Kbps.
While overprovisioning is laudable,
[EMAIL PROTECTED] wrote:
Hi all,
I have been facing about the problem to know who is online with asterisk PBX.
However users wanted to see it right away, without launching any application.
As I could not find any solution with IP phones and users were really
complaining, I decided to write this
AHBLWEB wrote:
That's good and will get me the account#.
I guess it's too much to ask for the same kind of tone prompts that occur
after dialling the 77 (boop-boop) and then after the 7 digits
(beep-beep-beep) before the user gets the final real dial tone?
If you want another dial tone, set up
AHBLWEB wrote:
Our current ROLM switch uses two-digit Feature Access Codes (FACs) for
long-distance calls to force the entering of a set of seven digits
representing a client and matter number before giving a real dial
tone. This sequence is passed as part of the CDR record and is used
to
Steve Underwood wrote:
Hi Angel,
It is working pretty well. I think it will be available about the end
of the year. I will not be free. It will be supplied with a
commercially licenced Asterisk.
Here's a question: if the author has purchased a commercial license to
use Asterisk, and I get
Benjamin on Asterisk Mailing Lists wrote:
However, I did not sign the disclaimers because it also asks for any
future rights. That's really stupid. If you were to invite me to a pub
and offer to buy me a pint of Guinness, then would it not be
unreasonable of me if I said, I accept, but only on the
Henry Devito wrote:
I know I am top posting and that is a no no, but I would like to
comment on
this generally. I just did this with a historic building with the same
situation cat3 two pair in each office. I used a Tut systems solution
called expresso this gave us cable TV and Ethernet to each
Michael Welter wrote:
We have a 100 year old building here in Colorado that needs a new
telephone system. The building (five floors) is steel frame with lath
and plaster walls. There is no crawl space above the ceilings or under
the floors. The building is historic, and nothing can be done to
Michael George wrote:
I'm having some trouble with DISA() in a call plan that worked before 1.0. If
anyone has experience with it, I would appreciate some advice.
Perhaps you could post relavent sections of your dialplan...?
___
Asterisk-Users
[EMAIL PROTECTED] wrote:
Folks,
I am trying to determine the best way to allow a station to monitor the
status of another station.
For example:
a reception set needing to see the status of 20 or 30 phones
OR
an executive assistant wanting to have appearances of several other
extensions, in order
Marconi Rivello [EMAIL PROTECTED] wrote:
Hi, I have a curiosity: how much does a regular PBX system cost? I'm
curious if using IP telephony in a building is cheaper than a regular
PBX, because of the high cost of the IP phones.
Take a look at
Is there a good place to find Asterisk consultants?
There is an asterisk-biz mailing list.
Nick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi. Just for grins, reverse the BNC's. You could have the
tx and rx crossed, happens all the time on ds-3's. Also,
with the BNC's unplugged and hanging, You should have a
Yellow light on the Hdsl Box (I've always called that a
smartjack, but we call football soccer, so what do we know?
Is it possible to have the system outdial and take surveys. either by
receiving DTMF or voice?
Yup. Just have the system use the outgoing queue (see sample.call) and
have it call an AGI script upon answering.
If you want CDR data, be sure you connect to an extension that starts the
AGI.
Hi Michael,
Michael Welter wrote:
the codecs I use? Filter-out everything between, say, 55 and 65Hz?
Notching may not be that effective, as it will not deal with the
harmonics. The analogue to digital converter should already be
filtering below 300Hz, so you probably have quite a lot of
Dear to all
someone know how is possible to have a DTMF tone like C AKA Alpha
Tone
(connect tone) to the caller?
Yes, it's possible.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hey Everyone,
Thanks for helping with this integration, I finally got it working!
I have inbound and outbound dialing from/to the asterisk box via a pri
line! I would definitely like to contribute this information to the
wiki so we can fill in some of the information gaps there. In the
Digium X100P / new cards are is available on ebay for $43.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3073050567category=3309
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3073050567category=3309
Hope this helps to who want to play with X100P! Are these being sold by
Digium ? I
Hi there,
I stumbled on this list mostly by accident. I came across Asterisk *
as a means to help me get a better handle on my soaring telephone
costs. Each month I look at my phone bills and my stomach just turns
because I can not find any competition to Verizon which is the local
WipeOut wrote:
Asterisk would need some kind of clustering/load balancing ability
(Single IP system image for the IP phones across multiple servers) to
be truely Enterprise Class in terms of both reliability and
scaleability.. Obviously that would not be as relevent for the analog
hard
On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote:
Yes, I've played with it a bit. It's pretty simplistic... the
clustering just keeps several servers in sync with each other. I
suppose that would be easy to do with Asterisk, especially if
configuration data was stored in a RDBMS that could
Andrew Kohlsmith wrote:
I would set the Enterprise Class bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
To turn around, let's discuss what we need to
On Thu, 18 Dec 2003, Aaron Martin wrote:
I have upgraded my grandstream phone from firmware 1.0.3.78 to
10.0.4.30 and now I am having problems with early dial. On the older
firmware earlydial worked fine with my asterisk server, but now as
soon as I have dialed the number I get a congested
Andrew Thompson wrote:
- Original Message -
From: Burak Balasaygun [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 27, 2003 10:46 PM
Subject: RE: [Asterisk-Users] Help with x101P
snip
I'm not sure what you mean by what type of switch you are connected to?
The
Brian West wrote:
Today class we are going to be talking about the wonderful line of
GrandStream products. Or should I say BarbieTone phones?
OK, so GrandStream phones are crap. What other phone products are there
on the market that are cheap (and I DO NOT want to buy phones off eBay
for a
Andrew Thompson wrote:
- Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: Asterisk User List [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 4:06 PM
Subject: [Asterisk-Users] 911 settings.
I would like to know if anyone has come up with a script for 911 dialing
rules that
Peter Pauly wrote:
Does anyone know what would be involved in making
Asterisk work as a voicemail system in a Centrex
environment? We have a Centrigram voicemail system
that belongs in the Smithsonian. There are analog
lines coming into the box and a 56KB data feed from
the phone company's
Michael Devenijn wrote:
Everythings works great with asterisk exept one feature with redirect
: it doesn't redirect when ringing ...
Have you used astman with a new CVS? It works for me...
If not, you'll need to post more information for the list to help you.
Nick
Adam Hart wrote:
Hence why I ask for a company name. Small correction to your post, if it's
distributed to anyone, the source must be available to EVERYONE.
IANAL, but I don't think that's quite accurate. If this person wanted
to, they could only ofter an offer for the source to people who
What kind of stability / reliability are people currently experiencing
with the Linux / Asterisk combination? We will be running 3-10 SIP
phones from India to US using nothing more than regular cable / dsl
connections from both locations.
People have had months of uptime. I would be more
Jonathan Moore wrote:
Quoting Nick Bachmann [EMAIL PROTECTED]:
Having a DSS (the blinking lights for each extension, short for
Digital Station Selector) is a feature that I wish Asterisk had. A
week or so ago there was discussion about a new Windows-based Asterisk
application (Asterisk
Senad Jordanovic wrote:
Nick Bachmann wrote:
Hello
I have couple of Grandstream phone and some of them after a day or
two just stops receiving calls, you can still make a call from that
phone but you cannot receive calls until you restart the phone. Is
it a wrong configuration of phone
Greg Boehnlein wrote:
First and foremost, these Key System installers are big believers
in VoIP and convergence technologies. While the KSU vendors may see
This has been my experiance as well. Everybody but PBX vendors like
VoIP. The KSU people like it because it gives them more work and
Ariel Batista wrote:
I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive!
1 - Sipura SPA-2000
2 - Grandstream Sip phone
Steven Sokol wrote:
1. Redial
2. Voicemail Box Monitoring
3. Enhanced Conferencing
4. Outlook/Act/Goldmine Integration (PIM stuff)
5. Call History (both inbound and outbound)
6. Redirect Option on Ring (VM, Application, Transfer, etc.)
7. Automatic mixing and delivery of monitored (recorded)
Steven Sokol wrote:
I have looked at creating a Console version of the application. It
would be very much like a DSS (Direct Station Selector for the
non-ATT/Avaya initiated). It would support either click-to-transfer or
drag-and-drop transfer of incoming calls.
Excellent! This is one
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