From: David Ruggles [EMAIL PROTECTED]
Date: Sun, 18 Feb 2007 20:41:46 -0500
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a
difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
extension number so most of the dialplain suggestions offered so far
You can just install CentOS with RAID and whatever you want, then use
the Trixbox tar package instead of the ISO. Still, why on earth did
the Trixbox team didn't leave the option of doing a custom install
with the ISO ?
/snip
Please note that the recent (2.x) releases of trixbox allow you to
At 20.13 18/02/2007, you wrote:
You can just install CentOS with RAID and whatever you want, then use
the Trixbox tar package instead of the ISO.
Very useful information. Thanks!
Still, why on earth did
the Trixbox team didn't leave the option of doing a custom install
with the ISO ?
I
Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet for an answer to this one. My phone
company
provides no disconnect supervision. Hence I'm forced to use the
busydetect
feature. I have a TDM400 with two FXO
Thanks Luki, that's exactly what I was looking for, I'll give it a try...
Regards,
Ricardo.
Luki wrote:
someone please give me one example?
[locals]
exten = _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten = _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten =
Hi List,
I am having some trouble with installing the latest version of ztdummy on a
CentOS Kernel 2.6 system.
I have installed a few Asterisk systems on Slackware Kernel 2.4.x without
any issues, unfortunately there is no choice about this distro, or kernel as
it has been preinstalled by
Asterisk and a modem pool.
I have a small modem pool, after its carry for an asterisk modems have
ceased will incorporate or incorporate, but for the speed 9600. I use
Tormenta 2b2.
Prompt the decision.
___
--Bandwidth and Colocation provided by
Hi All,
Is it possible to use asterisk as a internet link backup callback solution? I
mean when my main DSL link is down at my server room I would like to dial to
asterisk , then it will call back me and provide a connection to a LAN
network.
Regards,
Dominik
You can have multiple control channels on a PRI.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
On Sun, 18 Feb 2007, Matt wrote:
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have
all
the intelligence you need to do it right.
You may not have that option. For example, you
You mean compiling raw tar.gz or SRPMS? And where do you download
them from? Trixbox site or the original vendors' sites?
I just download the tarball from asterisk.org and compile it. Trixbox
is not a special version of Asterisk, it is just an easy way to
install Asterisk, FreePBX, FOP and a
Yes, it is complicated. We moved all the telco trunks to PRI last week
monday, but the configs for the old settings are still there. We will
be moving all the channels going into the Nortel onto a Nortel T-1 card
(nor PRI, PRI for Nortel requires a costly feature activation code) soon.
HI
I've configred an Incoming DID in my asterisk and when I call from outside I
see call is coming to my Asterisk server and then from asterisk it rings on
a particulat exten but when I pickup the call the call get disconnect
immediate and on the other end it keep trying (ringing).
here is my
I don't think Asterisk plays a role in this (unless I'm missing your point).
A simply script to ping your server room will do. Upon failure, the script
could initiate a PPP connection outbound.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dear all,
I've searched the web about Asterisk with Radius integration for user
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's
Radius client patch, an still open branch of Digium Issue Tracker SIP
peer authentication on an
From: Stephen Bosch
Hi, Trevor:
Trevor G. Hammonds wrote:
Stephen Bosch wrote:
Are BRI circuits what phone companies call digital lines for use
with digital sets, such as with digital Centrex?
I'm not aware that Telus even offers BRI.
Sorry -- BRI is ISDN, not digital Centrex.
I, too, have heard about that best practice of using different
channels for different AP's on the same SSID. As far as I can tell,
This is standard textbook stuff. Read Cisco press's 'Deploying License
Free Wireless Wide-Area Networks' by Jack Unger.
it's BS. I don't know who started it, but
The WAP54's have a 'repeater' mode which I've used on occasion.
Which is all well and good, but they use WDS which doesn't work with WPA.
Not on the WAP54's anyway (I learned the hard way on that one). Some
vendors have working solutions:
From what I've found, if you modify the normal config files, then they
will stay until the next time you update your server. When you update
your server, they get restored back to the origional, packaged state.
It's best to stick to the _custom files as much as possible.
John McCollough
LAN
Hi all,
I make others tests.
Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2
It works only if I use canreinvite= yes.
But all my clients are behind a Nat without ALG or stun stuffs...
Do you know if canreinvite= yes it's the only way to make it works??
Thanks a lot for your
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator transfers that
call to another employee, the Caller ID doesn't seem to do what I would
expect it to. I would expect it to show the original caller's ID.
Example:
John calls
I played around the wink and rxwink settings. While increasing rxwink does
delay the answer it still sees the DNIS digits individually. I changed the
signalling to featd and now I get the following error:
WARNING[27630]: chan_zap.c:5661 ss_thread: Got a non-Feature Group D input
on channel 1.
Not sure about others, but on Polycoms a blind transfer sends
original callerid, screened sends operators callerid
On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator
Not sure about an Asterisk only call transfer but in an Asterisk/SER
environment the SER server will use the REFER method to perform the
transfer. In this case ehe caller ID needs to be the contents of the
Refer-By header of the SIP message. Not the contents of EXTEN
-Steve
Jerry Jones
I probably have a screened transfer setup. Is that just a setting
somewhere I can easily change? I'm trying to avoid making users press
extra keys, like #1 or anything like that.
Rob
Jerry Jones wrote:
Not sure about others, but on Polycoms a blind transfer sends original
callerid, screened
A T.38 fax call typically begins as a normal voice media call. The call then
dynamically switches over T.38 image media on detection of fax handshake
tones. The dynamic modification of session from audio to image is
accomplished through SIP RE-INVITE messages. I would imagine canreinvite=
flag
Not an asterisk setting. It is how the endpoints perform the transfer.
On Feb 19, 2007, at 9:21 AM, Rob Schall wrote:
I probably have a screened transfer setup. Is that just a setting
somewhere I can easily change? I'm trying to avoid making users press
extra keys, like #1 or anything like
did u try modprobe zaptel first ? also check makefile if ztdummy is marked
for compilation or not
On 19/02/07, Chris Blunt [EMAIL PROTECTED] wrote:
Hi List,
I am having some trouble with installing the latest version of ztdummy on
a CentOS Kernel 2.6 system.
I have installed a few
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 16, 2007 12:56 AM
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote:
I have an ITSP provider that will only deliver calls using SIP
registrations (would prefer delivery to static IAX or SIP url, but hey),
periodically their servers don't respond to a renew request, and when
this happens the sip stack in asterisk (1.4.0) stops working until
either a SIP reload
Thank You all, thank you very much
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
i've installed trixbox with TE110P TDM400B, but no led is ON in the
TE110P, i don't know why even if the 4 leds of My TDM are greens
any explaination
Thank You
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asterisk-users mailing
--
Message: 2
Date: Mon, 19 Feb 2007 13:03:05 +0200
From: Eugeniy Khvastunov [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk and a modem pool.
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=KOI8-U;
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 19 Feb 2007, at 15:01, Noah Miller wrote:
The WAP54's have a 'repeater' mode which I've used on occasion.
Which is all well and good, but they use WDS which doesn't work
with WPA.
Not on the WAP54's anyway (I learned the hard way on that
On Mon, Feb 19, 2007 at 09:08:41PM +0530, Mail list wrote:
Once it is installed I run: modprobe ztdummy with the following result.
did u try modprobe zaptel first ? also check makefile if ztdummy is marked
for compilation or not
No. the module iwas not found. 'modinfo ztdummy' won't show
Yes, the canreinvite means Re invite, but there is a consequence in Asterisk
configuration.
For sure, all the signalisation traffic will go through the asterisk … but for
the RTP traffic?
If canreinvite = No, all RTP traffic will go through the Asterisk (useful for
NATed phoned without
RS == Rob Schall [EMAIL PROTECTED] writes:
RS Example: John calls in from the outside using (213-555-1234) and
RS he calls into the asterisk system (actually the operator). The
RS operator (a real person) answers the call and presses transfer on
RS her polycom 501 phone. I see an incoming call
I have setup an asterisk based phone system using snom-320 (SIP based)
phones.
I would like to change what seems to be the default procedure for an
attended call transfer. Right now, the phone user places the call on hold,
calls the extension using a extension button on the phone, speaks with
is it a spinlock problem? try
http://www.trixbox.org/modules/newbb/viewtopic.php?viewmode=flattopic_id=1626forum=2
- Original Message -
From: Mail list
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, February 19, 2007 3:38 PM
Subject: Re:
Without seeing your config files my guess would be that this is
something to do with a bad codec negotiation.
I'd bet that your IAX phone is using ulaw and your DID provider is using
something else like G729.
Mark
On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
HI
I've configred an
Hi all,
I have a bunch of Cisco 7941's with SIP firmware 8.2.1 that work great
on a LAN with Asterisk 1.2.x but fail miserably when trying to use them,
through NAT, with a public Asterisk server.
We have a bunch of Polycoms and Aastra phones that work great with NAT
so I'm very familiar with NAT
MB == Michael Boers [EMAIL PROTECTED] writes:
MB I have setup an asterisk based phone system using snom-320 (SIP
MB based) phones.
MB I would like to change what seems to be the default procedure for
MB an attended call transfer. Right now, the phone user places the
MB call on hold, calls the
Hi list,
I bought two UTStarcom F1000 phones, pre-equipped with the latest
firmware, including WPA support. Those are configured to register to an
asterisk server on the internet (not LAN), and registration works.
Calling and being called also, with transfer and all bells and whistles.
After a
Hey Guys,
I'm curious if there's an interest in a free, CallerID database? For those
of you in the same spot we are, our current provider only provides us with
the CND, excluding CNAM.
Would creating a public database, managed by users be worthwhile to anyone?
Thanks - Any input is greatly
Hi All
I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura
2000) and a CG-410 Gateway to connect the the two PSTN lines that I have.
I have a odd hassle that for no apparent reason, the calls will quite
working.
but quite I mean the phones will ring but their is no voice
Hey Shane,
The basis of my idea was that it would be user-moderated/generated. A
'owner/operator' of a number, would submit verify their phone number,
enter their caller id, and basically be done with it. The logistics of it I
don't really think would be that complicated. If a listing needs to be
Hello,
i've installed trixbox with TE110P TDM400B, but no led is ON in the
TE110P, i don't know why even if the 4 leds of My TDM are greens
any explaination
Thank You
No LEDs on TE110P (and similar cards) can sometimes mean that the Zaptel
driver isn't running. Can you run zttool and see
Greetings folks,
I'm currently dealing with a company to let me set Caller-ID-Name on
outbound calls. So far pretty happy with their services. The basic service
works like this:
* CLEC sets Point Code to point to this company
* CLEC has to sign LOA saying they give me permission to set the
I think terms of service for most CNAM providers prohibits sharing the
data and limits the amount of time it can be cached for your own reuse.
A public database managed by users would be of little value unless there
are means to verify the data. If people trusted it, outbound
telemarketers might
On 2/19/07, Robert Norton - SophMedia LLC [EMAIL PROTECTED] wrote:
Hey Guys,
I'm curious if there's an interest in a free, CallerID database? For those
of you in the same spot we are, our current provider only provides us with
the CND, excluding CNAM.
YES!
Would creating a public
Am Montag, den 19.02.2007, 12:39 -0700 schrieb Robert Norton - SophMedia
LLC:
Hey Guys,
I’m curious if there’s an interest in a free, CallerID database? For
those of you in the same spot we are, our current provider only
provides us with the CND, excluding CNAM.
Would creating a public
Our CNAM provider claims to have more than 196 million entries. I
just don't think you could reliably maintain that in this format.
Let's say I'm a CLEC and I have 40,000 numbers. I want to update that
in one place (my SCP, probably). I wouldn't also want to update
another database
Wireless wrote:
looks good.
9 says type
[EMAIL PROTECTED] ~]# modprobe zaptel
which returns nothing... when I run 10
At this point, if you run dmesg, do you find the following in your
kernel log?
Digium High-Performance Echo Canceller, version 8.20
Optimized for i386 CPU architecture
You MUST account for fraud, as well.
Perhaps proving you own the number, as in the LNP process, by providing the
cover page of the bill...
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Monday, February 19, 2007 3:02 PM
To: Asterisk Users Mailing
It seems like an interesting idea, but if this would be a public user
updated 411, who would ensure that was more up to date than 411. If the
numbers are off from 411, then the phone provider isn't keeping the
records properly. A customer should be notifying the phone company when
they are moving,
At least the current system involves some qualification(CLEC status) and
some policing(regulators can revoke CLEC status). Why create a system
where setting your CNAM requires about as much validation as registering
a domain name? You need to consider that most of the free data from
sources like
Jerry,
Thats a really interesting question... I have (aparrently) identical 7912
phones and some stay registered correctly for every while others drop out
and display the X.
I've never spent the time (yet) to investigate it further, but as they are
all running the same version of SIP
Michael J. Tubby G8TIC wrote:
Thats a really interesting question... I have (aparrently) identical
7912 phones and some stay registered correctly for every while others
drop out and display the X.
If you guys are using DHCP for your phones, make sure that your lease
time isn't too short. It
Hello,
Can anyone recommend the 'best' kernel and zaptel versions to use with
asterisk?
we're currently running trixbox and are having numerous call quality
issues(disconnects, echo, garbled speech) and I'm considering wiping the
asterisk box and installing a virgin copy of centos,
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul wrote:
I think terms of service for most CNAM providers prohibits sharing the
data and limits the amount of time it can be cached for your own reuse.
I don't know why they manage to get this level of control over the cnam database
so that they
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 19, 2007 8:21 PM
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote:
looks
My understanding is that with Asterisk 1.2.x issuing the command of make
install-udev allowed the drivers to be loaded upon the server boot.
Doing this with version 1.4 does not seem to work.
Using menuselect I selected zaptel and ztdummy. Should I also be
selecting something else for the
Why install Centos -- its really old? Check with whoever is supplying
your telephony hardware and see what kernel versions are needed or
will work with that hardware.
on Monday 02/19/2007 mail-lists([EMAIL PROTECTED]) wrote
Hello,
Can anyone recommend the 'best' kernel and zaptel versions
First my two cents. I dont think creating a system to store the info
is a good idea mostly because of the reasons mentioned in this thread.
An easy way to do free CNAM lookup, keep one pots line around with
caller id service on it for each non cached number that comes in make
a phone call to the
I use the mediatrix 1204 although not easy to configure it does an excellent job
On 2/19/07, Barry Fawthrop [EMAIL PROTECTED] wrote:
Hi All
I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura
2000) and a CG-410 Gateway to connect the the two PSTN lines that I have.
I have
There is an issue when using call-limit for a SIP interface in
sip.conf. The call count does not properly reset when some calls
end. The problem happens regardless of which side of the connection
ends the call. It happens on all calls including calls from SIP
interface to SIP interface (with
C F, your method is technically feasible. I thought of it over a year
ago but never posted it because it would definitely be considered an
abuse by most providers. If a lot of people start using it, I expect we
would start seeing TOS for PSTN and voip alike to start changing. Of
course the best
Does google really have the true CNAM database? When I enter my number,
I get a search result for my business listing at yellowpages.com
Are you referring to something available in a google area other than the
search engine?
Brad Templeton wrote:
On Mon, Feb 19, 2007 at 03:01:50PM -0500, Paul
I doubt it's CNAM since it has old an outdated listings.
On 2/19/07, Paul [EMAIL PROTECTED] wrote:
Does google really have the true CNAM database? When I enter my number,
I get a search result for my business listing at yellowpages.com
Are you referring to something available in a google area
My better guess is, that since Google is a web search engine all their
results are based on web gatherings, which is where Google is taking
the number resluts from.
On 2/19/07, Paul [EMAIL PROTECTED] wrote:
Does google really have the true CNAM database? When I enter my number,
I get a search
This compile error started happening about 2 weeks ago with zaptel.
/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c: In function ‘debugfs_open’:
/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.c:171: error: ‘struct inode’
has no member named ‘u’
make[4]: *** [/mysrc/asterisk/zaptel-1.4/xpp/xbus-core.o]
I have filed a bug report on this issue since it does work in 1.2.9.1 but not
in 1.4.0 and I tried the latest SVN and it still doesn't work.
If anyone else can try to reproduce this and see if they get the same error it
would be appreciated.
The dialplan should have
we're currently running trixbox and are having numerous call quality
issues(disconnects, echo, garbled speech) and I'm considering wiping the
asterisk box and installing a virgin copy of centos, compiling asterisk
myself and installing freepbx on it's own..
Is there anyone who can
Instead of forwarding to IAX softphone if I'll play some music same thing is
happening in this case also.
On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote:
Without seeing your config files my guess would be that this is
something to do with a bad codec negotiation.
I'd bet that your IAX
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns SIP/2.0 404 Not Found
any ideas ?
On Mon, Feb 19, 2007 at 11:03:17PM -0500, Noah Miller wrote:
You can probably solve quality issues without doing a complete
reinstall, and you should be able to get reliable results with most
all kernel versions and asterisk versions. There are some exceptions,
but versions of asterisk and
On Mon, Feb 19, 2007 at 10:50:28PM -0500, Robert La Ferla wrote:
This compile error started happening about 2 weeks ago with zaptel.
How about looking at my answer to your exactly same question from 5 days
ago?
--
Tzafrir Cohen
icq#16849755
On Tue, Feb 20, 2007 at 11:05:31AM +1100, Klaverstyn, David C wrote:
My understanding is that with Asterisk 1.2.x issuing the command of make
install-udev allowed the drivers to be loaded upon the server boot.
Doing this with version 1.4 does not seem to work.
Those udev rules are
I am running CentOS 4.4.
You say I need modprobe ztdummy on startup. I though the udev option
made that happen.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, 20 February 2007 3:11 PM
To: asterisk-users@lists.digium.com
On Sun, 2007-02-18 at 14:05 -0500, Gary H. Thompson wrote:
Hi,
[snip]
I understand what a broad scope I am asking about so would appreciate
any tips to help me get started. Since there are many ‘brands’ of
Linux what is the best one to start with? Which Linux will be better
when I get to the
I solved the problem by making 2 trunks from The Cisco Call Manager
using another port 5062 instead of 5060
Also at Asterisk I added a new SIP ...
And I modified the extension.conf with these :
exten = 558,1,Answer
exten = 558,2,Playback(message.wav)
exten = 558,3,Dial(SIP/[EMAIL
On Tue, Feb 20, 2007 at 07:37:02AM +0100, Patrick wrote:
On Sun, 2007-02-18 at 14:05 -0500, Gary H. Thompson wrote:
Hi,
[snip]
I understand what a broad scope I am asking about so would appreciate
any tips to help me get started. Since there are many ‘brands’ of
Linux what is the best
Ricardo Carvalho wrote:
Dear all,
I've searched the web about Asterisk with Radius integration for user
authentication, and got a bit confused...
I see that there have been some work around it, there is PortaOne's
Radius client patch, an still open branch of Digium Issue Tracker SIP
peer
create user trunk on each box and dialplan to make call
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns
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