Hi,
I've just discovered that dahdi_genconf could create configuration files
reading a single genconf_parameter file.
1. Is using dahdi_genconf recommended to configure dahdi or is it a helpful
tool for specific devices or installation processes ?
2. If not, is there a custom mecanism that
Hi all,
i want to configure free radius with asterisk server. my radius server and
clients are running perfectly with each other. now only thing i need to do
is to patch radius client with asterisk.
Can any one suggest how i can do that? i am using free radius 2.1.x version
and asterisk latest
On Mon, Dec 01, 2008 at 08:39:53AM +0100, Olivier wrote:
Hi,
When typing man dahdi_genconf, you can read :
This script generate configuration files for Dahdi hardware. Currently it
can generate three files: dahdi, chan_dahdi, users and chan_dahdi_full
where it say :
This script
Hi,
Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2
Scheme:
1--
| | -- cat5 patch cord
On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
Hi,
I've just discovered that dahdi_genconf could create configuration files
reading a single genconf_parameter file.
1. Is using dahdi_genconf recommended to configure dahdi or is it a helpful
tool for specific devices or
2008/12/1 Tzafrir Cohen [EMAIL PROTECTED]
On Mon, Dec 01, 2008 at 08:39:53AM +0100, Olivier wrote:
Hi,
When typing man dahdi_genconf, you can read :
This script generate configuration files for Dahdi hardware. Currently
it
can generate three files: dahdi, chan_dahdi, users and
2008/12/1 Tzafrir Cohen [EMAIL PROTECTED]
On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
Hi,
I've just discovered that dahdi_genconf could create configuration files
reading a single genconf_parameter file.
1. Is using dahdi_genconf recommended to configure dahdi or is it a
On 1 Dec 2008, at 09:09, Olivier wrote:
Is there something obvious in configuration keeping this self-
looping trunk to work ?
You are not using a crossover cable?
___
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2008/12/1 Tzafrir Cohen [EMAIL PROTECTED]
On Mon, Dec 01, 2008 at 09:12:31AM +0100, Olivier wrote:
Hi,
I've just discovered that dahdi_genconf could create configuration files
reading a single genconf_parameter file.
1. Is using dahdi_genconf recommended to configure dahdi or is it a
2008/12/1 Steve Howes [EMAIL PROTECTED]
On 1 Dec 2008, at 09:09, Olivier wrote:
Is there something obvious in configuration keeping this self-
looping trunk to work ?
You are not using a crossover cable?
No : B410P manual says using jumper is enough (but I've got no successful
experience
On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
Hi,
Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2
Scheme:
1--
It looks like you are trying to dial out on your 'NT' instead of your
'TE'.
Try changing Dial(DAHDI/g1/${EXTEN:1}); to Dial(DAHDI/G1/${EXTEN:1});
Oh, and I'd use mISDN for BRI as DAHDI always gave me problems.
HTH
___
-- Bandwidth
2008/12/1 Tzafrir Cohen [EMAIL PROTECTED]
On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
Hi,
Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2
Scheme:
2008/12/1 Andrew Thomas [EMAIL PROTECTED]
It looks like you are trying to dial out on your 'NT' instead of your
'TE'.
Try changing Dial(DAHDI/*g*1/${EXTEN:1}); to Dial(DAHDI/*G*1/${EXTEN:1});
Unfortunately, changing g1 to G1 didn't change end result.
Anyway, I can't really see why using
Apart from you were dialling out on your inbound context and
vice-versa.
The best advice I can give now is to change to mISDN - as this
is proven to work with v1.4 and v1.6.
Actually - have you tried putting the 100ohm termination on for
your NT
On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
Hi,
Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2
Scheme:
1--
Hello,
I'm working with asterisk 1.6. And I have success using func_odbc with one
row query results (SELECT source,destination from cc WHERE ... ):
exten = s,1,Ringing
exten = s,n,Wait(4)
exten = s,n,Answer
exten =
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using func_odbc with
one row query results (SELECT source,destination from cc WHERE ... ):
exten = s,1,Ringing
exten = s,n,Wait(4)
exten = s,n,Answer
exten =
2008/12/1 Tilghman Lesher [EMAIL PROTECTED]
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using func_odbc with
one row query results (SELECT source,destination from cc WHERE ... ):
exten = s,1,Ringing
exten = s,n,Wait(4)
On Tue, 2008-11-25 at 08:06 +, Grey Man wrote:
On Mon, Nov 24, 2008 at 6:56 PM, Steve Murphy [EMAIL PROTECTED] wrote:
For the moment, let's not worry about the implementation. Let's
get consensus on the spec first. In the scenario, where A calls B,
B xfers A to C, C xfers A to D, or
On 1 Dec 2008, at 13:38, Giedrius Augys wrote:
2008/12/1 Tilghman Lesher [EMAIL PROTECTED]
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using
func_odbc with
one row query results (SELECT source,destination from cc
WHERE
Max Alex schrieb:
Actually we are setting up the callerid in case of emergency calls when we
got the anonymous callerid from the caller.
But the calls are going with callerid anonymous and not set the callerid, i
want to know how can we sent some meaning ful information to the emergency
On Wed, 2008-11-26 at 01:13 +0100, Freddi Hansen wrote:
To me the obvious answer is to provide a CDR for every call leg so for
A calling B via Asterisk there would be two CDRs produced. It's far
far easier to disregard the unwanted CDRs than it is to try and
generate the missing ones
Hi murf,
Speaking as someone who designs and builds billing platforms, this is
very exciting.
One little thing I have most problems with is the good old fax
detection. I know that NVFaxDetect et al do actually answer the call
and, therefore, get flagged as ANSWERED in the CDR.
But, if the call
Steve Murphy wrote:
[...]
Everyone--
I've just made some major changes to the CDRfix2.rfc.txt
file in http://svn.digium.com/svn/asterisk/team/murf/RFCs
to accommodate the Leg approach instead of a
channel-based approach.
Greyman is correct. By cutting the timeline into legs,
we
Just seconding Freddi's idea - as it makes perfect sense. Otherwise we
could quite easily start testing a call that hasn't actually finished
yet.
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asterisk-users mailing list
Steve Murphy wrote:
On Wed, 2008-11-26 at 01:13 +0100, Freddi Hansen wrote:
To me the obvious answer is to provide a CDR for every call leg so for
A calling B via Asterisk there would be two CDRs produced. It's far
far easier to disregard the unwanted CDRs than it is to try and
JD schrieb:
Steve Murphy wrote:
Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID set
has been closed. If so, then the entire set is finished. We can use this
facility to get you a closing attribute, that could be
...or something along the lines of a setting a variable (like we do for
MONITOR_EXEC)...
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To UNSUBSCRIBE or update options visit:
2008/12/1 Tzafrir Cohen [EMAIL PROTECTED]
On Mon, Dec 01, 2008 at 10:09:06AM +0100, Olivier wrote:
Hi,
Hardware is b410p with :
- span1 jumpers set to NT (no 100 ohm termination)
- span2 jumpers set to TE (no 100 ohm termination)
- straight cable from port1 to port2
Scheme:
Hello,
Groups in asterisk are summarized here (
http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
(as I've been advised in another thread, to switch from one notation to the
other and I can't see the reason behind
On Mon, 2008-12-01 at 10:55 -0600, JD wrote:
Steve Murphy wrote:
[...]
I love it! You will have it done later today, correct? (joke.)
Just a non-technical/social suggestion: don't call this CDR. Call it
Enhanced CEL or something like that.
Why?: because otherwise you will forever get
Olivier wrote:
Hello,
Groups in asterisk are summarized here (
http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and DAHDI/g1/0123456789
(as I've been advised in another thread, to switch from one notation to the
other and I can't
2008/12/1 Andrew Thomas [EMAIL PROTECTED]
Apart from you were dialling out on your inbound context and vice-versa.
The best advice I can give now is to change to mISDN – as this is proven to
work with v1.4 and v1.6.
I wanted to try the Digium B410P card with Dahdi as those are now
On Dec 1, 2008, at 9:07 AM, JD wrote:
Steve Murphy wrote:
Freddi--
Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID
set
has been closed. If so, then the entire set is finished. We can use
Olivier wrote:
Hello,
Groups in asterisk are summarized here
(http://www.voip-info.org/wiki/view/Channels+and+Groups).
Is there any difference between DAHDI/G1/0123456789 and
DAHDI/g1/0123456789 (as I've been advised in another thread, to switch
from one notation to the other and I
When system is busy, asterisk uses 99.9% cpu.
I want asterisk to use more 100% cpu to process more calls.
Am I in a time warp? Is this April 1 in disguise?
The difference between 99.9% and 100% is not discernable. The other .1% is
probably used determining that asterisk is using the rest.
Thanks for replying !
I've added a link here
http://www.voip-info.org/wiki/view/Channels+and+Groups to
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels.
2008/12/1 Mark Michelson [EMAIL PROTECTED]
Olivier wrote:
Hello,
Groups in asterisk are summarized here
http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
FUD? Interesting? Boring? New news? Old news?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice together?
Mark Bergen
Information Systems Manager
Number TEN Architectural Group
Winnipeg - 204.942.0981
Victoria - 250.360.2106
www.numberten.comhttp://www.numberten.com
___
--
Sounds possible, but as a user of uTorrent, I have yet to see this feature
It may simply be that I havnt looked hard enough.
I can say, that I still have to have a tcp port routed for uTorrent to work
properly.
I may post an update, If I notice a change in this behavour.
--Christopher Dobbs
On Dec 1, 2008, at 9:07 AM, JD wrote:
Steve Murphy wrote:
Freddi--
Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID set
has been closed. If so, then the entire set is finished. We can use
this
Daniel Hazelbaker wrote:
On Dec 1, 2008, at 9:07 AM, JD wrote:
Steve Murphy wrote:
Freddi--
Very interesting. Brian Degenhardt had some code we just gave some
thought
to, wherein we determine if the last channel involved in a linkedID
set
has been closed. If so, then the
We have one connected.
What's your question ?
On Monday 01 December 2008 13:49, Mark Bergen wrote:
Anyone familiar with getting Asterisk 1.4 and Mitel 3300 to play nice
together?
Mark Bergen
Information Systems Manager
Number TEN Architectural Group
Winnipeg - 204.942.0981
Victoria -
The DAHDI docs actually include the 'steps' that used to be in Zaptel's
'make b410p'. These steps involve downloading and compiling mISDN. Why
re-invent the wheel?
Just a thought :-).
As for the 100ohm termination bit - it's simply changing a couple of dip
switches on the b410p card (as
JD schrieb:
As to the idea of piping to a deamon via socket or dbus: how would
asterisk behave if the daemon froze or worse, it lagged?
Asterisk could write the CEL events to the database and either
on every event or after some kind of final event send a signal
to the socket, i.e. write a
SPAM SPAM SPAM SPAM[0]
Please:
1. Try to use a decent mailer that does not break threading.
2. Don't reply to messages you get with the extra word [SPAM] in the
subject. This breaks thing even worse, as even the subject line is
changed.
Sanity has been restored to the subject.
Back to your
Try to use a decent mailer that does not break threading.
This is an opportunity for me to ask a question regarding this mailing list.
I've worked with several other groups using a variety of communications
techniques from Web based to news reader based, but never anything like this.
Due to
Setup is:
/etc/dahdi/system.conf:
span=1,0,0,ccs,ami
bchan=1,2
hardhdlc=3
span=2,1,0,ccs,ami
bchan=4,5
hardhdlc=6
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
[channels]
context=isdntrunk
switchtype=euroisdn
group=1
immediate=no
signalling=bri_cpe
channel=1-2
group=2
signalling=bri_net
Can some tell me what this warnings means?
The dialplan works, but I get these warnings every once in a while:
Log:
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Identifier 0,
identifier_type 2 not found in identifier list
[Dec 1 12:26:31] WARNING[20425] app_addon_sql_mysql.c: Invalid
2008/12/1 Andrew Thomas [EMAIL PROTECTED]
The DAHDI docs actually include the 'steps' that used to be in Zaptel's
'make b410p'. These steps involve downloading and compiling mISDN. Why
re-invent the wheel?
in 1.6.1rc5, mISDN is optional ...
Just a thought J.
I will compare both ASAP
On Monday 01 December 2008 02:43:17 pm Wilton Helm wrote:
Try to use a decent mailer that does not break threading.
This is an opportunity for me to ask a question regarding this mailing
list. I've worked with several other groups using a variety of
communications techniques from Web based
Barton Fisher wrote:
Can some tell me what this warnings means?
The dialplan works, but I get these warnings every once in a while:
I'm guessing that some times the caller-id is blank. I got tired of
those errors and did the following before the query:
exten =
Wilton Helm schrieb:
This is an opportunity for me to ask a question regarding this mailing list.
I've worked with several other groups using a variety of communications
techniques from Web based to news reader based, but never anything like this.
Due to my lack of experience (and/or
Never heard of that mailer. You might try using Kontact under Kubuntu, as it
has reasonable defaults.
Outlook Express, the default mail handler on Windows XP. I only have one Linux
computer at present and it is dedicated to Asterisk and file backups and web
serving, etc., so whatever I use
Barton Fisher wrote:
Can some tell me what this warnings means?
The dialplan works, but I get these warnings every once in a while:
I'm guessing that some times the caller-id is blank. I got tired of
those errors and did the following before the query:
exten =
Tilghman Lesher schrieb:
On Monday 01 December 2008 02:43:17 pm Wilton Helm wrote:
I am using
[...]
OE 6 for E-Mail
Never heard of that mailer.
:-)
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied -
Philipp Kempgen schrieb:
And in most MUAs or servers you can define a filter / processing
rule so you can store mails matching a certain rule (
List-Id header contains asterisk-users.lists.digium.com or
To contains asterisk-users@lists.digium.com or
Sender contains [EMAIL PROTECTED])
in a
On Mon, Dec 01, 2008 at 01:43:17PM -0700, Wilton Helm wrote:
Try to use a decent mailer that does not break threading.
This is an opportunity for me to ask a question regarding this mailing
list. I've worked with several other groups using a variety of
communications techniques from Web
Please help.
Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
You can use next parameter:
Fromuser = VoipDirect777821
At 04:23 p.m. 01/12/2008, Shaun Wingrin wrote:
Please help.
Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=
accountcode=5260477782
amaflags=billing
You shouldn't open text your password. Shouldn't IP on Asterisk 2 be
1.2.3.4?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin
Sent: Monday, December 01, 2008 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound calls from
Thanks. I have them going to their own folder now which is a great help. It
doesn't appear that OE can organize them in the folder, though. I does a good
job in a news folder, but doesn't seem to have that ability in a mail folder.
Nor does it create a usable To field for replies. I still
Hi All,
I'm testing the Asterinic Call Center Queue Log Analizer. Working ok
except for realtime monitoring. The page updates queue summary and
calls waiting, but not Agent status. When an agent is (busy) in
[asterisk queue show], the 'state' of the agent in agent status on
the web page does
At 12:34 12/1/2008, Alex Balashov wrote:
http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/
FUD? Interesting? Boring? New news? Old news?
Hmmm. When our users are pounding the network
with BitTorrent traffic, we just shut them down
and wait for them to complain. It's
Doug wrote:
Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me. We tell
our customers that they are not allowed to
download copyrighted material. But for other,
legal BitTorrent transfers, we suggest that
they use the scheduling feature of uTorrent to
RE Kushner List Account wrote:
Doug wrote:
Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me. We tell
our customers that they are not allowed to
download copyrighted material. But for other,
legal BitTorrent transfers, we suggest that
they use the
Alex Balashov wrote:
RE Kushner List Account wrote:
Doug wrote:
Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me. We tell
our customers that they are not allowed to
download copyrighted material. But for other,
legal BitTorrent transfers, we
Awesome, I always wanted to see this law in real life.
Thank You!!
Alex Balashov wrote:
RE Kushner List Account wrote:
Doug wrote:
Why the BitTorrent guys want to give themselves
even a worse reputation is beyond me. We tell
our customers that they are not allowed to
download
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to download public domain
material? That kind of restricts the amount of information
available on the Internet. Nitpick:
Igor Widlinski wrote:
Awesome, I always wanted to see this law in real life.
Technically I didn't call him THAT guy. I was thinking of that recently
elected Chicago street thug who speaks before large crowds at night.
Just spend ten seconds on YouTube and you'll see it's not my original
I know I can setup asterisk without Internet at all and it works as
local pbx.
Would an asterisk box work with a dynamic IP, with a dyndns name?
What must I take care if I try that?
bye
Ronald
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RE Kushner List Account wrote:
The question is, what are you actually paying for as a customer? To
discriminate against bits just because they actually use what they are
paying for is beyond me.
At least a bandwidth cap is easier to understand. You get what you pay for.
Speaking as a
Alex Balashov wrote:
RE Kushner List Account wrote:
The question is, what are you actually paying for as a customer? To
discriminate against bits just because they actually use what they are
paying for is beyond me.
At least a bandwidth cap is easier to understand. You get what you
BJ Weschke wrote:
Alex Balashov wrote:
RE Kushner List Account wrote:
The question is, what are you actually paying for as a customer? To
discriminate against bits just because they actually use what they are
paying for is beyond me.
At least a bandwidth cap is easier to understand.
Hi,
How can I park a call from dialplan and get going??
Example:
1. Answer
2. While follow = false
3. ParkCall
4. Checksomthing à follow = true
5. Endwhile
6. UnParkCall
7. Go on
..
The idea is let the call waiting while I do
We're using it here on dynamic IP from our ISP.
They provide reverse DNS, which we've simply setup a CNAME to.
So, CPE390480Q239432098423.MYISP.COM is cnamed to PBX.MYBUSINESSDOMAIN.COM
Did not have to change anything else for this to work.
Thanks,
Matt G
: http://www.voipphreak.ca
:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to download public domain
material? That kind of restricts the amount of
The Asterisk.org development team has released Asterisk versions 1.2.30.3,
1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1
and 1.6.1-rc2. These releases are available for immediate download from
http://downloads.digium.com/.
This update for Asterisk includes a
2008/12/1 Tim Panton [EMAIL PROTECTED]
On 1 Dec 2008, at 13:38, Giedrius Augys wrote:
2008/12/1 Tilghman Lesher [EMAIL PROTECTED]
On Monday 01 December 2008 06:15:15 Giedrius Augys wrote:
I'm working with asterisk 1.6. And I have success using func_odbc with
one row query results
Hello,
Now I'm testing func_odbc and hash. My configurations are:
func_odbc.conf
[GETNUMBER]
dsn=sqlserver
;mode=multirow
;rowlimit=10
readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers
WHERE number=${SQL_ESC(${ARG1})}
extensions.conf
exten = s,1,Ringing
exten =
Hi,
Testing latest 1.6.1, it occurred to me I had to add a couple of noload
statements in /etc/asterisk/modules.conf to remove ERROR messages, when
starting Asterisk.
(I don't imply those ERROR messages were fatal to Asterisk but as a general
rule, I tried to start Asterisk without any of those).
Hi,
As latest asterisk-libpri-dahdi is introducing dahdi support of B410P, can
we use High Performance Echo Canceling addon with B410P ?*
Regards
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