Andrew Gillham wrote:
Senad Jordanovic wrote:
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to
be of quite good quality. However placing calls from ATA is not
possible at all to any
Billy Huddleston wrote:
change dtmf to info on both * and in the handytone.
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 8:01 PM
Subject: [Asterisk-Users] Handytone 286 - calling out
Hi,
Just received
What is your H.323 configuration? You must provide
the contents of oh323.conf and the relevant lines of extensions.conf
(preferably off-list).
Michael.
Antonio Sanz wrote:
Hi,
First at alll, I beg your pardon because maybe I explained bad my
questions (because my low level english)
I have
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I didn't
want to get told off for not looking first)!
I don't have enough Asterisk boxes (yet) to test this scenario.
If had 3 Asterisk boxes,
Asterisk wrote:
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I didn't
want to get told off for not looking first)!
I don't have enough Asterisk boxes (yet) to test this scenario.
If had 3
Hi all,
We are about to make our first channel bank install. This will be a one
PRI outside connection and up to 70 extensions.
As the schedule (and the budget) is pretty tight, I would like to learn
a little bit more about general experiences with channel banks, like
echo cancellation
Regards,
Christopher J. Wolff, VP CIO
Broadband Laboratories, Inc.
http://www.bblabs.com
Hello,
I'm running a bone stock * box that only has SIP clients and about a dozen
cisco T1 gateways. Some of the higher delay users complain that they
occasionally hear echo on local and long distance
Juha is correct about the SC600.
I picked up a 2.4ghz P4 SC600 for $270 with KB and mouse and free shipping.
if you have time to wait, they have some super deals. This dude has
probably the best Dell deal website around.
http://www.rasputinj.com/cat3.html
Thanks,
Steve T
- Original
Hi,
I have received some replies for my previous mail (* configuration), asking
for my goals in configuring Asterisk. So here they are:
We are planning to host an Inter continental virtual PBX service that will
enable our users to register for an account which give them a toll-free # or
a
On Tue, Nov 25, 2003 at 05:51:37PM -0600, Don Pobanz wrote:
Modem connect speeds on calls through * seem to be lower than calls
made through the telephone company lines or our old Rolm PBX. All data
calls have 2 wire analog modems on both ends.
For my set up I have channels of a Zhone
Girish Gopinath wrote:
Hi,
I have received some replies for my previous mail (* configuration),
asking for my goals in configuring Asterisk. So here they are:
We are planning to host an Inter continental virtual PBX service that
will enable our users to register for an account which give them
How should I solve this ?
Did nobody recognize this problem ?
I se it as a major bug, or am I doing something wrong ?
/Mike
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I recommend using Adtran 750's. You can get them setup with all your FXS
ports on Ebay for around $ 500.00 used but in great shape! These work great
out of the box. If your only using 70 extensions you will need 3 of these
unit's. In our setup we have 4 Adtran 750's one 600 and incoming PRI and
An end user contacted us who needs a fairly basic database driven IVR system
written. It looks like AGI would work for his situation.
If any of the resellers on the list do contract AGI scripting, please
contact me off list so that I can put you in touch with him.
Sean
Does anyone know if a web interface has been created for * ?
--
*
Not everyone is touched by an Angel
Those that are, never forget the experience
*
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I've not had ANY problems using info OR rfc2833.. I did have problems using
inband. Try switching to it and see how it works.. I NEVER had a problem
with double digits, and, I believe that the reference to GS phones having
that problem with * was retracted.
Thanks, Billy
- Original Message
I am having some issues when trying to connect with perl to the asterisk
manager and doing an IAX2 show channels.
If i do that on a server that is heavily loaded, i sometimes get some
events instead of the channels i asked for.
Any suggestions how i could fix that ?
zoa.
On Wed, 2003-11-26 at 04:27, Asterisk wrote:
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I didn't
want to get told off for not looking first)!
I don't have enough Asterisk boxes
This is totally awesome. I was scouring the net for a long time for
something like this and could find very little if anything. This is an
exceptional find. Thanks for sharing it with us!!
-edwin
-Original Message-
From: Jonathan Biggs [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
I've just been informed, in an IRC room, that it is possible to use a
modem card with *. Can someon please confirm this for me? Thanks in
advance.
--
*
Not everyone is touched by an Angel
Those that are, never forget the experience
*
Does it work through the command line? telnetting to your Asterisk box to
the Manager interface?
In you perl script are you using Net::Telnet to connect? or the
Asterisk::Manager?
I have had several issues with the Asterisk::Manager module so I switched to
Net::Telnet and it works every time.
I am a bit puzzled about the meaning of the different jitter buffer options.
If i set:
Dropcount=3 what effect will this have exactly ? (will this have an
influence on how fast a jitter buffer is built or destroyed?)
jitterbuffer=200 - will this create a fixed buffer of 200ms this implying
On Wed, 26 Nov 2003, Angel Gabriel wrote:
I've just been informed, in an IRC room, that it is possible to use a
modem card with *. Can someon please confirm this for me? Thanks in
advance.
why don't you also mention that after being informed of this, about 10
different people also told you
Hi WipeOut,
Thanks for the information provided. I would be greatful if u can
clarify the following doubts:
Girish Gopinath wrote:
..
..
Now, lets assume I have a linux box on a dual processor 3.2 GHz Intel
box with 4GB RAM, RAID system and at the data center we would have
On Wed, 2003-11-26 at 10:22, Girish Gopinath wrote:
1) How many Voicemails can be recorded at a time?
Depends on where the call is coming from (PSTN via Tx00P or VoIP) and
what format you are storing voicemails in (wav or gsm)..
***
What difference does it make for PSTN and VoIP?
A PSTN
Steven Critchfield wrote:
On Wed, 2003-11-26 at 04:27, Asterisk wrote:
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I
didn't want to get told off for not looking first)!
I don't have
- Original Message -
From: Angel Gabriel [EMAIL PROTECTED]
To: * Users [EMAIL PROTECTED]
Sent: Wednesday, November 26, 2003 10:55 AM
Subject: [Asterisk-Users] Modem cards??
I've just been informed, in an IRC room, that it is possible to use a
modem card with *. Can someon please
Just a php config file interface. check out phpconfig in cvs. its just for editing
and
parsing the conf files
Dave
[EMAIL PROTECTED] 11/26/2003 7:33:04 AM
Does anyone know if a web interface has been created for * ?
--
*
Not everyone is touched by an Angel
Those that are,
Hi!
If SIP/U2 transferred the call to an extension that made use of the
switch statement... What would the call path be?
Would the call traffic go from A1 in A2 back out of A2 to A3?
...or would it be switched and go directly from A1 to A3?
The theory - as far as I was able to find out -
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
answer, busy, etc etc
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Chris Albertson wrote:
One other idea might be to use the USB FXO interface but I don't
know how well it works. Some peole here have complained about
sound quality
We use sip fxo gateways via enet.
Works great.
We go in and give demos on their phone lines.
--- costas [EMAIL PROTECTED] wrote:
I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card.
Using
DIAX as softphone and dialing out to PSTN generally results in good
sound
quality at softphone end (no echo), but PSTN end experiences quite a bit
of
echo. I have enabled echosquelch in capi.conf, but it does not
Hi all,
Is anyone here using linux as a router and managing their VoIP traffic with
CBQ ? If so, do you have any configs (tc scripts etc) to share? We've trying
to ensure that all VoIP traffic is prioritised ahead of 'normal' traffic,
and at the moment have setup two classes based on the TOS
did you configure dialpeer voip in the cisco .
pointing to * ip addr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Pavel
Litvinenko
Sent: Tuesday, November 25, 2003 10:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco to use * as a gateway?
On Wed, Nov 26, 2003 at 09:26:18AM -0800, Richard Lyman wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I came up with this header record after looking through the
I'm having major problems routing a modem (data) call through my
Asterisk box. I've got a single incoming POTS line through my X100P,
and a few extensions (and modems) plugged into the ports on my TDM400P.
I've been using the system for a few weeks for voice applications and
everything is
On Wed, Nov 26, 2003 at 12:12:06PM -0600, Walker Haddock wrote:
On Wed, Nov 26, 2003 at 09:26:18AM -0800, Richard Lyman wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
Thanks for the truly useful feedback. I'm having a real hard time with
the FAQ pages listing
RH 8 9 FIRST in the list of Linux distros that Asterisk compiles and
runs on and having
any bugs (oh I mean RH problems) discarded. It would be much more help
to have responses
such as yours or to
On Wed, Nov 26, 2003 at 01:55:36PM -0500, Clif Jones wrote:
Thanks for the truly useful feedback. I'm having a real hard time with
the FAQ pages listing
RH 8 9 FIRST in the list of Linux distros that Asterisk compiles and
runs on and having
any bugs (oh I mean RH problems) discarded. It
On Wed, 2003-11-26 at 12:55, Clif Jones wrote:
Thanks for the truly useful feedback. I'm having a real hard time with
the FAQ pages listing
RH 8 9 FIRST in the list of Linux distros that Asterisk compiles and
runs on and having
any bugs (oh I mean RH problems) discarded. It would be much
On Wed, Nov 26, 2003 at 06:18:00PM +0100, Philipp von Klitzing wrote:
The theory - as far as I was able to find out - involves:
transfer=yes/no in iax.conf
canreinvite=yes/no in sip.conf
I have run into a related problem, and can't figure my way
out of it. This involves two asterisk
Oh its been tested with DB2, MySQL, Text Files and PostgreSQL... Works
like a charm! :P
bkw
On Tue, 25 Nov 2003, Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=578
Just in case anyone else wants more instructions. :)
bkw
On Wed, 26 Nov 2003, Asterisk wrote:
I'M
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I've done a search in Google, I even went to dictionary.com to check the
meaning of the word, but I don't know why it always equals 4 in my
Angel Gabriel wrote:
Does anyone know if a web interface has been created for * ?
There's one in the source tree, phpconfig. There's a lot of other projects
out there, depending if you want
* A configuration interface
* A user interface
* A receptionist/manager interface
See
Thanks for the quick feedback! I don't have a lot of free time to play
with Asterisk
right now but a friend of mine wanted me to get it working on Red Hat
for him which
resulted in the RH problems/questions. Personally, I prefer Debian
which suites my
needs for embedded projects and hacked up
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
/O
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--- [EMAIL PROTECTED] wrote:
On Wed, 26 Nov 2003, Angel Gabriel wrote:
I've just been informed, in an IRC room, that it is possible to use
a
modem card with *. Can someon please confirm this for me? Thanks
in
advance.
why don't you also mention that after being informed of this,
Just found this IETF draft, that gives a lot of examples on how to implement
PBX features in SIP. Good inspiration!
http://www.voip-info.org/tiki-index.php?page=SIP+PBX+functions
/Olle
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On Wed, Nov 26, 2003 at 08:33:13PM +0100, Olle E. Johansson wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I've done a search in Google, I even went to dictionary.com to
Hello!
*sigh*
Is it wrong to reply to your own post?
Thanks for the info guys. After I'd read you replies I looked at another
post I made... the answer to this question was answered in README.cdr
(of course)!
:)
Ben
__
Benjamin Wakefield
[EMAIL PROTECTED]
Hi,
Anyone
know anything about Asterisk's support for door phones? Receiving the call from
the door intercom system, opening the door, etc?
Any
hardware recommendations? I understand that the equipment we have now is Panasonic
proprietary and came with the currently deployed Panasonic
On Nov 26, 2003, at 10:34 AM, Peter Zeltins wrote:
I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card.
Using
DIAX as softphone and dialing out to PSTN generally results in good
sound
quality at softphone end (no echo), but PSTN end experiences quite a
bit
of
echo. I have
On Thu, Nov 27, 2003 at 12:44:58AM +0300, Shoval Tomer wrote:
Hi,
Anyone know anything about Asterisk's support for door phones? Receiving
the call from the door intercom system, opening the door, etc?
Any hardware recommendations? I understand that the equipment we have
now is Panasonic
Hi,
Anyone know anything about Asterisk's support for door phones? Receiving
the call from the door intercom system, opening the door, etc?
Any hardware recommendations? I understand that the equipment we have
now is Panasonic proprietary and came with the currently deployed
Panasonic
a. thanks, looks great.
b. can anyone recommend a solution that costs a little under a 1000$ ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock
Sent: Thursday, November 27, 2003 2:05 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition
for?
I've done a search in Google, I even went to dictionary.com to check
the
meaning of the word, but I don't know why it always equals 4 in my
CDR!
I tried to convert a cheapie box with a VIA C3 processor into my
Asterisk server with a TDM-400P rev. E. It didn't work. :) I'm just
posting my experiences here for the record; this is not a plea for
assistance.
The nifty-looking blue E cards, as most here probably know, require a
PCI
Tom Shoval wrote:
a. thanks, looks great.
b. can anyone recommend a solution that costs a little under a 1000$ ?
See Door Entry Systems at www.vikingelectronics.com
Jorge
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock
Sent: Thursday,
I have a client who needs an application for there field techs to call
in when they arrive on site and when they leave. The logic behind it
seems pretty simple. I am going to write something in AGI to capture
some DTMF tones and update this data into MySQL to run some reports
from.
But here's
I think I figured out my callerid issue... For some reason the callerid
is not getting passed anymore... This is controlled by the PhoneCompany
- Yes?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Wednesday, November 26, 2003 7:22
Or that asterisk is answering too soon to get the callerid, it is
transmitted between the 1st and 2nd ring.
Plug in a normal phone to the line, call it and see if it gets the
caller id.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
I actually have it running over an 802.11b link to the front gate entrance.
So if I can inject RTP or IAX2 frames I can open your
front door? Cool! keyless entry!
;)
--
/~\ The ASCII Ribbon Campaign
\ /No HTML/RTF in email
X No Word docs in email
/ \ Respect for open standards
Tom Shoval wrote:
a. thanks, looks great.
b. can anyone recommend a solution that costs a little under a 1000$ ?
See Door Entry Systems at www.vikingelectronics.com
my solution uses a $12 walmart phone, and a $20 door strike, you can't get
much less expensive than that. If the posted
I called the Phone company they have messed up the line.. They are
working to resolve it..
But I am trying to figure out how to collect the DTMF tones in AGI
(STDIN). The application keeps hanging up on me, before I can enter any
keys. I have tried adding the wait_for_digit.. But I am not
Door phone resources:
http://www.vikingelectronics.com/
Door phones, elevator phones, and other interesting telephone equipment
Their door entry systems are here:
http://www.vikingelectronics.com/products/doorentry/product_list.html
http://www.homephone.com/
home of the
Dear All,
We are all Windoze C++ developers but are working
on C/C++ development for linux. But we needsomeone of our teamto be
trained quite soon in Asterisk development to add some features we really need
inside the company.
We are searching somebody or a company who will
train a developer
[EMAIL PROTECTED] wrote:
Door phone resources:
http://www.vikingelectronics.com/
Door phones, elevator phones, and other interesting telephone
equipment Their door entry systems are here:
http://www.vikingelectronics.com/products/doorentry/product_list.html
Would
Hi,
Anyone know anything about Asterisk's support for door phones?
Receiving the call from the door intercom system, opening the door,
etc?
Any hardware recommendations? I understand that the equipment we
have now is Panasonic proprietary and came with the currently
deployed Panasonic
Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server?
rdgs,
/Staffan Kerker
No.
There are currently requests in the system for that functionality
(http://bugs.digium.com/bug_view_page.php?bug_id=134) but it's
waiting for a White
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish to test and or make this better?
(I know some of the code can be put into functions)
bkw
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[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
Anyone know anything about Asterisk's support for door phones?
Receiving the call from the door intercom system, opening the door,
etc?
Any hardware recommendations? I understand that the equipment we
have now is Panasonic proprietary and came with the
Surely dick smiths have something
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, 27 November 2003 4:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] door phone
[EMAIL PROTECTED] wrote:
Hi,
I recently setup an Asterisk Server-
I was able to follow a tutorial from
http://www.automated.it/guidetoasterisk.htm#_Toc49248752
Until it told me to call another line, let it ring until voice mail picks up.
My problem is the tutorial left out how to configure a SJPhone so that it connects to
I know there are several folks hanging around the list that resell VoIP
products, I'd be interested in getting your info off-list as I'm trying to
persuade someone to get me a budgetone 101 for Christmas ;) I've checked
the yahoo site, and they seem to only carry it in white.. Ideally, I'd
like
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