See if it helps you:
exten = _0.,1,Dial(Zap/1/${EXTEN:1})
exten = _7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
If you want to call FWD# 5 then you dial: 75. If you want to
call 911 then: 0911.
You must have registered to FWD in the sip.conf.
If there are other extensions starting with 7
Hi,
I have a dlink dvg-1120M (mgcp) box that i will like to use with
asterisk. Is it possible? has anyone done that?
Here's a link to the product page at dlink.
http://support.dlink.com/products/view.asp?productid=DVG%2D1120M
Also, does anyone has or know where to get the firmware for
The Cisco 7905G phones can be mounted on a wall quite easily. They
also support PoE (Cisco PoE;
http://www.voip-info.org/tiki-index.php?page=Cisco%20POE might be
useful).
-Shaun
- Original Message -
From: David Gomillion [EMAIL PROTECTED]
Date: Fri, 3 Sep 2004 16:41:29 -0500
Subject:
hi,
i have some newbie questions about channel banks. i have an adtran
act-1241 sitting around. it accepts D4 modules, and it contains a number
of em cards.
first of all, how does this thing work? a t1 contains 24 channels, and i
noticed that the channel bank has space for 24 cards. what do
On Sun, 2004-09-05 at 03:10, Ilia Mirkin wrote:
hi,
i have some newbie questions about channel banks. i have an adtran
act-1241 sitting around. it accepts D4 modules, and it contains a number
of em cards.
first of all, how does this thing work? a t1 contains 24 channels, and i
noticed
A channel bank allows you to go from DS1 to DS0; i.e. it takes the 24 channels
from a DS1 (T-1) and spins off individual DS0's.
For instance, you could plug a T-1 PRI (23 channels + 1 channel for signaling)
into a channel bank and get DS0 POTS (plain old telephone service) lines, which
can
On Fri, Sep 03, 2004 at 10:26:58AM -0700, Paul Mahler wrote:
The Mepis Debian distro is pre-configured for *, www.mepis.org They spent a
lot of time making Mepis work with * out of the box.
In what ways (comparing to the Debian packages)?
--
Tzafrir Cohen
On Sat, 2004-09-04 at 22:53, Paul Mahler wrote:
Asterisk should run well with any Linux distribution. Mepis,
www.mepis.org, is pre-configured for * and might make your
installation faster and easier.
Paul
Can you elaborate on what you mean by pre-configured.
Thanks Marconi,
That's pretty much what I'm doing, although I'm using IAX. I've been over
the file several times and can't spot what I'm doing wrong...no use of the
. seems to work properly. My guess is that I've made a mistake somewhere
else in the file. Nothing without a fixed number of
[tovpc]
exten = _81XX,1,Macro(dialvpc,${EXTEN:1},70)
Change the above to one X with a . after it. Right now, it will accept 10
digits, no more, no less. Putting the period in
makes that 81 plus at least 1 digit until we timeout looking for more
digits.
New line:
exten =
On Sep 3, 2004, at 4:47 PM, Chris Shaw wrote:
Ok Way OT, I didn't mean to get into a religious debate, I like the
Intel
cards, I have several of them and recommend them to my friends, etc...
Be that as it may... This was using these cards in a software bridge...
significantly more traffic than
Thanks Lyle,
I tried that...now it let's me dial two digits after the 7 (i.e. 761 ring
busy) and two digits after the 81 (i.e. 8161 ring busy)
The log shows Call rejected by IP: No such context/extension
Any idea why it won't wait for more digits, even with the 7X. ???
On 2004-09-05 10:18,
Hello,
Could anybody suggest cheap FXO/FXS devices with full T.38 support over
SIP?
I found a number of devives with declared H323/SIP and T.38 support but
some of them supports T.38 only with H323, others have buggy T.38
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
I just got a Cisco 7940G phone running SIP firmware version POS3-06-1-00. I
unlocked the phone's config, and using the soft keys, I entered the SIP
Configuration menu and keyed in values for Name, Auth. Name, ProxyAddress
(where I gave my Asterisk server's IP address), etc.
The result is that I
Also, in case it makes any difference, I have the PhoneJack card in
dialtone mode in /etc/asterick/phone.conf.
I can put it in immediate mode, but then when I pick it up I get no
dialtone and the log says it is sent into invalid extension 's' in context
'default'
The context specified in
Hi,
I'd like to test my links with remote locations. I wonder if there are any
echo asterisk server that could be called for quality estimation
Regards,
Robert.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Take a look at the options, DigitTimeout and ResponseTimeout in
extensions.conf. Those will fix the problem with not enough time to dial
the complete number.
Lyle
- Original Message -
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks - where do those go, is it a global option, or do I have to put it in
my internal context? Sorry if it's a dumb question, I'm very new to
asterisk...
On 2004-09-05 11:37, Lyle Giese [EMAIL PROTECTED] wrote:
Take a look at the options, DigitTimeout and ResponseTimeout in
Umar Sear [EMAIL PROTECTED] wrote:
On Sat, 2004-09-04 at 22:53, Paul Mahler wrote:
Asterisk should run well with any Linux distribution. Mepis,
www.mepis.org, is pre-configured for * and might make your
installation faster and easier.
Can you elaborate on what you mean by
I guess I am getting lost as I don't have the complete config files to go
through to verify everything. At least I started you off in the right
direction with some of the problems. I just looked through my conf files
again and in extensions.conf, the only place I put DigitTimeout and
Robert Rozman [EMAIL PROTECTED] wrote:
I'd like to test my links with remote locations. I wonder if there are any
echo asterisk server that could be called for quality estimation
You could sign up for FWD, using SIP or IAX2, and dial 613:
http://www.freeworlddialup.com/
--
_/
Thanks, I'm really stuck...perhaps it's something specific to the quicknet
phonejack card. No matter what I do, I can't configure a dialing pattern
for a phone plugged into the card unless the pattern has a fixed number of
digits.
Everyone tells me that exten = _9X.,1, ... Should get anything
All - NEED HELP BADLY..
Using Asterisk with H323 Channels..
Everything is installed but we cannot register w/ Lucent gatekeeper.
We ran ethereal and found that it was making GRQ (Gatekeeper discovery
requests)..
We had provided the name of the Gatekeeper (it's IP) and cannot determine
why
it's
Perhaps this will help...
I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I
get a dial tone. When I dial a certain number of digits, the call is
processed by Asterisk.
The question: How does Asterisk determine how many numbers to let me dial?
I'm banging my head
Huddleston, Robert wrote:
All - NEED HELP BADLY..
Using Asterisk with H323 Channels..
Everything is installed but we cannot register w/ Lucent gatekeeper.
We ran ethereal and found that it was making GRQ (Gatekeeper discovery
requests)..
We had provided the name of the Gatekeeper (it's IP) and
Everyone we have a few new things to give back to the asterisk community.
http://bugs.digium.com/bug_view_page.php?bug_id=0002379
http://bugs.digium.com/bug_view_page.php?bug_id=0002380
http://bugs.digium.com/bug_view_page.php?bug_id=0002381
These include app_chanspy, the ability to spy on ANY
BRAVO!
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, September 05, 2004 1:26 PM
Subject: [Asterisk-Users] ChanSpy by anthm and more...
Everyone we have a few new things to give back to the asterisk community.
Brian: Now the U.S. Law Enforcement guys will be happy.
Sorry but I could not resist the comment. Thanks it will help
people testing.
http://bugs.digium.com/bug_view_page.php?bug_id=0002379
http://bugs.digium.com/bug_view_page.php?bug_id=0002380
Hello All,
I have gone thru all the resources I could find on
google on asterisk + iconnect and managed to get outgoing calls working.
However,
I cannot get incoming calls to work at all. With
the sip debug on, I can see that something is happening everytime a call is
received
from
Am I doing something wrong? I can't get this dial command to timeout
Dial(Zap/g1/xxx,20)
--
Gary White [EMAIL PROTECTED]
Network Administrator Internet Pathway
105 D East Church Street Voice: 601-776-3355
P. O.
because of lack of answer supervision in *.
this is my problem too.
- shabanip
On Sun, 05 Sep 2004 12:49:40 -0500, Gary White (Network Administrator)
[EMAIL PROTECTED] wrote:
Am I doing something wrong? I can't get this dial command to timeout
Dial(Zap/g1/xxx,20)
--
Gary White
Also don't forget to visit us at Astricon... :)
Brian
Asterlink.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Sunday, September 05, 2004 12:26 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ChanSpy by
http://bugs.digium.com/bug_view_page.php?bug_id=0002384
Also res_sqlite is out... :)
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Sunday, September 05, 2004 12:57 PM
To: 'Asterisk Users Mailing List -
Thanks, Now I know it is not me and my config.
because of lack of answer supervision in *.
this is my problem too.
- shabanip
On Sun, 05 Sep 2004 12:49:40 -0500, Gary White (Network Administrator)
[EMAIL PROTECTED] wrote:
Am I doing something wrong? I can't get this dial command to timeout
Is there a Pause or Wait character for dial tone command for
the Dial command? Like in a modem Dial string? Im having issues with the
FXO Analog line not returning dial tone quick enough for * to recognize.
Arick
___
Asterisk-Users
Lowercase w for wait.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Arick Davis
Sent: Sunday, September 05, 2004 1:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Pause or Wait character in Dial command?
Is there a
Dial(Zap/g1/w18005551212)?
Arick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Sunday, September 05, 2004 11:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Pause or Wait character in
The problem you are having is due to the way chan_phone was designed.
The distributed driver does not buffer the entire phone number dialed
and then send it on to the PBX,
like a SIP phone would, but instead scans the dial plan after every
digit is entered to look for a match.
The solution is to
Please see my rely for the related topic
Eric Jacksch wrote:
Perhaps this will help...
I have a phone connected to a QuickNet PhoneJack card. When I pick it up, I
get a dial tone. When I dial a certain number of digits, the call is
processed by Asterisk.
The question: How does Asterisk
there IS answer supervision if you use a PRI or BRI - dunno about some
of the CAS options used in the US though. You don't say what type of
Zap channel you are using. If you are referring to an analogue call on
an X100P or similar, you will not get answer supervision unless you can
use a
Just to clarify the usage of the . wildcard in your dialplan.
Here is the proper usage of this feature which seems to not be documented
ANYWHERE very well.
[default]
include = other
exten = _712XXX,1,NoOp,Blah
[other]
exten = _7.,1,NoOp,somethingelse
The extensions in the current context win
Thanks Karl, very much appreciated...now I can stop smacking my head against
my desk :)
I pulled the source from CVS yesterday...assuming it's not in that, could I
get a copy of the new driver or a diff?
On 2004-09-05 14:50, Karl Brose [EMAIL PROTECTED] wrote:
The problem you are having
Not sure I understand..does that help my problem of not being able to enter
sufficient digits, or is that a consideration once I get a driver that
allows me to # terminate the dialing string?
On 2004-09-05 15:00, Brian West [EMAIL PROTECTED] wrote:
Just to clarify the usage of the . wildcard
Actually it does the proper usage of the . char in your dial plan should
solve this problem. It's not the channel driver that's doing this its
asterisk. You need to sandbox a wildcard into its own context then include
it. Otherwise it wins NO MATER WHAT. This way an extension defined within
Thanks Tim, I am using X100P cards on analog lines. I will try some of
you suggestions.
there IS answer supervision if you use a PRI or BRI - dunno about some
of the CAS options used in the US though. You don't say what type of
Zap channel you are using. If you are referring to an analogue
Does it removes the need of external databases (mysql, postgres) or it will
work with existing databases?
-Kannaiyan
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, September 05, 2004
Humm...I tried it and it didn't work, still rang after two digits, but I
may be making newbie errors :)
Should this work? Is NoOp a valid instruction, or do I need to create
something else?
[internal]
include = freeworld
Include = localexts
exten = _712XXX,1,NoOp,Blah
[freeworld]
exten =
Hi,
just a short question:
Can I use * with chan-capi to build up an internal S0 (for ISDN)?
Regards and tia Daniel
--
NEU: Bis zu 10 GB Speicher für e-mails Dateien!
1 GB bereits bei GMX FreeMail http://www.gmx.net/de/go/mail
___
Asterisk-Users
No it doesn't its just a nice standalone res that allows you to use SQLite
from the dialplan, cli and as a CDR engine and sqlite_Switch.
Its great for a standalone pbx because you can do something like this:
exten = s,1,SQL(SELECT total,balance,lastpaydate FROM customers WHERE
Latest version of res_perl is up also.
http://www.bkw.org/~brian/res_perl.tar.gz
Brian
Asterlink.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Sun, Sep 05, 2004 at 10:51:32PM +0200, [EMAIL PROTECTED] wrote:
Hi,
just a short question:
Can I use * with chan-capi to build up an internal S0 (for ISDN)?
No, cause chan_capi uses isdn-cards in TE mode.
You will need a card in NT mode to connect a isdn-phone to.
Therefor you need a
Tim,
Adding callprogress=yes to my zaptel.conf solved my timeout problem.
Thanks
--
Gary White [EMAIL PROTECTED]
Network Administrator Internet Pathway
105 D East Church Street Voice: 601-776-3355
P. O. Box 777
Here are the snippets...I changed things to 9 just in case...
No matter what I do, I get to dial 9 plus two more digits...
[internal]
;include = extensions
;include = tovpc
include = tofwd
; this should just dial myself
exten = _999XXX,1,Dial,${P1}
[macro-dialwfd]
exten =
And your using chan_phone?
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Jacksch
Sent: Sunday, September 05, 2004 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Wildcards
No Brian,
The old driver scans the ENTIRE dial plan on EVERY digit dialed so no
matter where, if you have a
. wildcard in the plan, it will match always on the first digit dialed.
It is the driver that does this.
If you use a SIP phone, or any technology that presents a complete dial
string,
Hello!
I want to use asterisk -rx show version from a php script called in
the browser using the local apache, which runs as user apache.
Asterisk is running as root.
I added the following line to /etc/sudoers using visudo:
apacheALL = NOPASSWD: /usr/sbin/asterisk
When i am on the
why not use a tcp socket and use the manager api and avoid the
permission issues all together
enable it in manger.conf and you connect over tcp log in and execute
the command nice and cleanly in your application. There should be
decent examples on voip-info.org
On Sun, 5 Sep 2004 23:52:13 +0200,
Salve,
I'm somewhat stuck on how to get DTMF working with my setup
and googling didn't yield anything similar.
My setup consists of one CAPI-capable board (AVM Fritz!DSL)
connected to a BRI (T-ISDN), one HFC-S board running in NT-mode
connected to an internal S0 bus with some ISDN devices (DECT
No, newer code does exactly how I described it. Specific matches in the
current context override wildcards in any included context. I have tested
this and that's how Mark himself says it works. This is how it should work
if I understand it correctly and I usually do, ast_matchmore_extension is
Just tried it and no luck here. Here's a copy of the script if you
want to go into details (note, I am having the same problem for all
scripts that came with Asterisk Manager. The error I get when trying
to execute in browser is: Server Error 500 - Premature end of script
header) I've spent 3-4
David Gomillion wrote:
I am looking for a large number (probably about 100 or so) low-cost
phones that I can hang on the wall. I need the phones to use PoE. Do
the Uniden phones support wall-mounting? These phones are not going to
be high-usage; they simply need to be there in case of an
Do not use sudo -u apache, that switches to the apache user and
runs the command /usr/sbin/asterisk -rx show version.
The asterisk command needs to be run as root, so your PHP script
would exec sudo /usr/sbin/asterisk -rx show version.
quote who=Roland Zagler
Hello!
I want to use asterisk -rx
On start I'm getting a warning (among other things)
chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny
disabled
Could that be making things worse?
I've reconfigured things (as per the snippets here) and it does seem to
match after each digit I dial...
Is there any new code
hi guys i'm having problem with my extention i dont have any actual phone huked up to
my asterisk but i was trying to use it as voice mail server and when i dial my
extention this is what i get from the terminal window {channel of type 'Console'
== Everyone is busy/congested at this time
so
On Sat, Sep 04, 2004 at 01:18:59PM -0500, Shekhar Prasad wrote:
Hi all,
I've installed Asterisk on Linux Red Had 9. Now, I was trying to set
up a GUI based system for the PBX.
I downloaded some packages, but I have to have Perl running CGI
scripts through the webserver. It does not allow
On Sun, Sep 05, 2004 at 11:52:13PM +0200, Roland Zagler wrote:
Hello!
I want to use asterisk -rx show version from a php script called in
the browser using the local apache, which runs as user apache.
Asterisk is running as root.
I added the following line to /etc/sudoers using visudo:
Amazing,
We are talking about CURRENT CVS CODE chan_phone the way it DOES work,
not zap, not anything else, not the way it SHOULD work.
CHAN_PHONE scans the ENTIRE dial plan on EVERY digit dialed
in dialtone mode and what you describe does not work for chan_phone.
Why is it so hard to accept the
Thanks to everyone for their help and comments on this. You've all been
very helpful. I've actually got outbound calls working on it fine right
now without having to change the configuration on the Mediatrix box at
all, as I don't have the Unit Manager Software at the moment. Outbount
seems
Do you have early dial enabled at all?
Craig
- Original Message -
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 06, 2004 5:16 AM
Subject: RE: [Asterisk-Users] Wildcards and variable number of
Well then chan_phone is broken and shouldn't take much work to fix it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Karl Brose
Sent: Sunday, September 05, 2004 7:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don't think so, but I'm very new to Asterisk - is there an easy way to
check?
On 2004-09-05 20:56, Craig Guy [EMAIL PROTECTED] wrote:
Do you have early dial enabled at all?
Craig
- Original Message -
From: Eric Jacksch [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Why do I even bother trying to help... I even pointed out that the channel
driver is at fault... I only pointed out how it should work and you get an
attitude about it GOOD JOB... Check out line 749 in chan_phone.c you'll see
and compare it to chan_skinny.c and see if maybe that fixes it(hint
Paul Mahler wrote:
The Mepis Debian distro is pre-configured for *, www.mepis.org They spent a
lot of time making Mepis work with * out of the box.
Erm the only issue with debian is they would have to do is mess with the
kernel module packages for digium hardware, even then it's not very
hard,
Zahid,
I can configure this box for Asterisk if you can put this unit on a Public
IP.
Seshu Kanuri
- Original Message -
From: Zahid Mehmood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Sunday, September 05, 2004 12:02 AM
Subject: [Asterisk-Users] need help
Thank you, but where I the command line would it be placed?
Arick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Sunday, September 05, 2004 11:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
Try welltech 3502 (2-port) or 3504A (4-port). beware it only works if
your 2 endpoints are not too many hops apart.
Elman Efendiyev wrote:
Hello,
Could anybody suggest cheap FXO/FXS devices with full T.38 support over
SIP?
I found a number of devives with declared H323/SIP and T.38 support but
Hi,
T.38 should be completely insensitive to the number of hops. That is its
whole reason for existing. It sounds like these units are not using T.38.
Regards,
Steve
Leo Ann Boon wrote:
Try welltech 3502 (2-port) or 3504A (4-port). beware it only works if
your 2 endpoints are not too many hops
Steve,
Correct me if I'm wrong, T.38 can be implemented with or without TCP. I
suspect these units only support T.38 over UDP, just like Cisco gateways
(see
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/faxapp/t38.htm).
In an ideal world, all fax relay
Could anyone who has successfully configured Asterisk to
use g729 to conference 10-20 people please post their configs. I purchased and
successfully installed 2 g729 licenses and but when I dial into my conference
number on the Asterisk box from a SPA-2000 set to allow all codecs, it always
Roger,
I haven't had any problems doing confs w/ g729. My guess is the Sipura
is asking for ulaw first. Try adjusting the codec priority on the
sipura side. IF you still have problems I an get my spa-3000 out and
trying and solve it for you.
-- William
- Original Message -
From: box100
Jamie Carl wrote:
Thanks to everyone for their help and comments on this. You've all
been very helpful. I've actually got outbound calls working on it
fine right now without having to change the configuration on the
Mediatrix box at all, as I don't have the Unit Manager Software at the
Bob Knight wrote:
There is a linux package called mbrowse that you can use with your
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not
spent
that much time on it.
I don't even have the MIBs which is half the
While I understand everything that you have said, I'm still a little
confused. Yes - I have what looks like a centronics connector on the
back. So, I can do t100p with em signalling - act-1241 em card
- what? Namely, if the EM card deals with the T1 end of the channel,
how do I get that to a real
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