Hi,
I used Cisco 3640 with 2xNM-HDV-2E1 cards.
The default GW router has RTP and TCP/UDP header compressions.
There is also a Linux solution for this. You can run RTP compression on
your asterisk box, and or run UDP/TCP header compression on the default
GW router.
Do you have a working * box at
I am not a coder hence this question:
If a web interface (similar to vonage account management) gets produced
using PHP/MYSQL to administer
*, does that require licence from Digium if the code is not open source.
Thanks...
Senad
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Asterisk-Users
While it seems this discussion is taking on here, i'll just jump in myself.
A couple of weeks ago i started playing with the idea of a webadmin, but one
that is bit different then the phpconfig there is now. Allthough i think
it's a good thing, don't get me wrong, i myself had something else in
Tjardick van der Kraan wrote:
While it seems this discussion is taking on here, i'll just jump in myself.
A couple of weeks ago i started playing with the idea of a webadmin, but one
that is bit different then the phpconfig there is now. Allthough i think
it's a good thing, don't get me wrong, i
Hi, I thought you were using * and was wondering which kind of PC server
you used to compress 120 voice channels.
Yes I have a working * (1xE1 PRI + analog)
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul Hakeem
Sent: Wednesday, October 01,
I've spent the last couple of days learning Doxygen and getting at least
basically familiar with the Asterisk source code. I'm starting to write
up comments for Doxygen to generate API docs from, and I've also started
looking at ways to use Doxygen to generate a configuration reference with
I have setup Asterisk to work with a SIP gateway, some SIP phones
and the Digium FXS/FXO development card combo on another *
box with pretty good results so far. Here are a couple of questions
that I have that wasn't obvious from the documentation:
Voicemail vs Voicemail2 - What is the major
Hello,
Could somebody tell me what I should change in
iax.conf file to be able to receive calls from iaxtel.
I am already registered and I can make calls to
IAXtel users but what I should do in iax.conf
to be able to receive call also.
-- Bart
Hi folks,
I'm still having the following problem, maybe someone can help me out of
it.
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
communicate through IAX2. Everything works ok on machine 1. On machine
2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I
According to [EMAIL PROTECTED]:
Does anybody have any thoughts on this plan, or better ideas?
Negative/positive thoughts about Doxygen?
Most importantly, is anyone else working on something along these lines
already?
Daniel,
Documentation is much-needed, and I'm glad to hear that there's
From the discussion thread, you will notice that the only real answer is to get a
lawyers opinion. And even there the final answer is the judge. So hope for the best
and be prepared for the worst.
It seems no one can agree on this. And even a lawyer would probably be confused. Why?
Because if
- Original Message -
From: Roderick Montgomery [EMAIL PROTECTED]
Date: Wednesday, October 1, 2003 10:46 pm
Subject: Re: [Asterisk-Users] Asterisk Documentation
... snip ...
integration with basic scripting languages, I'm not focusing at all on
documenting Asterisk code development.
I agree on the aspects of documentation.
However, my inset is that most of the basic functions (e.g. an example
dialplan) can be found on the net.
Since I am interested in building a powerful and usable VoIP gateway, I
would like to see more of the subtler and not so often exposed functions
Is anyone using nikotel with asterisk? When I attempt to place a call,
I get Everyone is busy at this time.
Thanks,
Kevin
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Subject pretty much hits it.
Can someone explain the difference?
Is global var / db similar? They effect all channels.
db lasts through restarts and vars are gone?
And Var is per channel?
John
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On Wed, 2003-10-01 at 07:56, costas wrote:
From the discussion thread, you will notice that the only real answer
is to get a lawyers opinion. And even there the final answer is the
judge. So hope for the best and be prepared for the worst.
Or if you wish to write closed source apps, you
If a web interface (similar to vonage account management) gets produced
using PHP/MYSQL to administer
*, does that require licence from Digium if the code is not open source.
If it merely manipulates Asterisk's config files and in no way links to
Asterisk itself, then the choice of licenses is
Matt,
It's done by using the switch keyword in extensions.conf
Thus if you fill in the stuff below correctly and make
the appropriate settings in iax.conf:
switch = IAX/username:[EMAIL PROTECTED]/context
Will send all extensions which cannot be resolved in the local dialplan,
over IAX to the
Nick Knight wrote:
Hello all,
I am sure that this is possible - for helpdesk envioroments i.e. when
you hear an announcment Your call may be recorded for quality purposes
can asterisk record all calls onto disk or similar - hopefully as MP3?
Thanks
Nick
great, thanks for that.
What if interface triggers CLI commands?
Senad
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* has application called Record.
type show appliaction record for more info at CLI prompt.
Senad
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Senad Jordanovic wrote:
* has application called Record.
type show appliaction record for more info at CLI prompt.
Senad
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Record is not the
Hello,
Sorry for posting again my question about MGCP Phone and Asterisk But I
can't use it.
I'd like to know weather it is a pb of my confiuration (mgcp.conf), My
IP Phone device or asterisk.
I include my mgcp.conf file and may send some debug trace.
Thank you for any feedback.
Best
yes, that is correct.
I just realised that myself.
Thanks
Senad
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hello,
I have been searching the mailing list for info on R2 signalling support
in *
The best I came up with was a post by Steve as far back as 18 Jul 2003.
This post indicates that some work was being done on it. Has anyone got
any info if the current cvs does have support for R2 on E100P
Could use some assistance please
I have failed to get past the cisco gauntlet and am prevented
from accessing the 7940 firmware required to get SIP on the phones I
purchased on ebay. I would be happy to buy the cisco support but they tell
me I have to be IP-tel certified first,
On Wed, 2003-10-01 at 10:41, rjrae wrote:
Could use some assistance please
I have failed to get past the cisco gauntlet and am prevented
from accessing the 7940 firmware required to get SIP on the phones I
purchased on ebay. I would be happy to buy the cisco support but they
On Wed, 2003-10-01 at 06:32, Paulo Mannheimer wrote:
Hi folks,
I'm still having the following problem, maybe someone can help me out of
it.
Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *)
communicate through IAX2. Everything works ok on machine 1. On machine
2, if I try to
No interrupts are being shared ;-(
take a look
CPU0
0: 155590 XT-PIC timer
1: 3 XT-PIC keyboard
2: 0 XT-PIC cascade
3:1534436 XT-PIC wcfxo
4: 49926 XT-PIC serial
7:1534528 XT-PIC
Does anyone know if there is a zapata.conf option to tell * to
listen for a dialtone before dialing?
I've got a couple of analog phones on a pstn line shared with a
x100p * fx line. If someone is on the analog phone and another
person initiates a call through * to use the same line, * dials
On Wed, 2003-10-01 at 10:23, Paulo H. Mannheimer wrote:
No interrupts are being shared ;-(
take a look
[snip]
My guess is that four Digium cards is probably too many for one box to
handle... they are probably overloading your system with interrupts.
(I'm just guessing at this point,
Rich Adamson wrote:
Does anyone know if there is a zapata.conf option to tell * to
listen for a dialtone before dialing?
I've got a couple of analog phones on a pstn line shared with a
x100p * fx line. If someone is on the analog phone and another
person initiates a call through * to use the
You need to specify what port and address its listening on and please
include your extensions.conf exten lines to use mgcp
/etc/asterisk/mgcp.conf
[general]
port = 2427
binaddr=192.168.0.1 ; if this is your asterisk box' ip
-Greg
- Original Message -
From: Daniel ANDRE [EMAIL
Thank you Greg,
I have fogotten to say that I can call my MGCP phone from my SIP phone
but mys MGCP Phone can't place a call.
Here is some parts of my extensions.conf:
[general]
static=yes
writeprotect=yes
[globals]
dandre = sip/p-dan.phone.iris-tech.fr
swiss1 = mgcp/aaln/[EMAIL PROTECTED]
Here's a good example: wiki? there is a wiki? :-) I saw a broken link
once but heard no more...does it still exist?
Oh yes, there is a Wiki.
And there's a lot of how to's and software documentation up there.
If you search the archives, I've mentioned the correct URL so often
that people propably
I was looking at some fixes in the replies to the chan_sip.c problems and
I am wondering if I am seeing the same thing in the earlier file version. I
just checked to see that my chan_sip.c is version 1.179 when I did my
checkout so I never had the later versions. The problem that I am seeing
is
I am trying to optimize echo cancellation. Originally (with only one
phone attached), echo cancellation worked well. Echo was only obvious
in the first second or so of a call. Now, with multiple phones on the
system, the echo does not go away. It is quiet, but audible and somewhat
annoying.
Take a look at the switch app..
Also search the archives for switch..
Thank you, I've missed that function.
Found an example in the archives and in the sample extensions.conf in the distribution.
Updated the wiki ;-)
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
Hello all
I new to this list and would like to ask a few questions with the following
assumtions.
1. im iggy to astrick
2. i havent made any dec as to what softswitch i will use
I am curious as to the quality of this asterisk softswitch and what success
or failure stories on deploying asterisk,
Are there any sip providers out there providing full
business telephone service. Not
just single line/residential service like I have seen with vonage etc.
For example take a company currently using a legacy pbx
connected to the PSTN with a PRI. I
would like to replace this setup with
We can do that in the UK and in Poland and soon in more countries, but since
you are US based I would recommend rather to talk to Jeremy at Nufone.
On Wednesday 01 October 2003 8:30 pm, David Harris wrote:
Are there any sip providers out there providing full business telephone
service. Not
Dave Weis wrote:
On Wed, 1 Oct 2003, Lists wrote:
Does anyone out there use Asterisk with voip(sip or iax) long distance
provider?
Care to share about your experiances doing this?
I'm using nufone.net. I just paypal'ed him some money, he sent me the
configuration changes and password info,
Hi George -- I think the answer is going to depend upon your extention
situation -- how many extensions do you plan on having?
- Jeff Dodge
- Original Message -
From: George L. Carden III [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 01, 2003 4:28 PM
Hello,
I am thinking about buying grandstream phones and I
have following question.
Does granstream ip phonesupport attendend
transer and blind transfer?
-- Bart
Only
blind transfer I think. Attended transfer can be sort of be done using asterisk
call parking feature.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Bartosz
JozwiakSent: 01 October 2003 20:04To: ASTERISK
USERSSubject:
An update on this issue:
It seems to be an echo cancellation problem. My setup is as following:
FXO - IAX2 server 1 - IAX2 server 2 - FXO
1st call has a terrible echo. If I start server 2 and immediately issue zap
destroy channel X (one of the 4 fxos), than the call goes through without a
Is anyone on the list using an Audiocodes gateway with asterisk and SIP?
I'm looking at that platform, but I have a couple of issues:
1) Echo cancellation. The echo that I'm hearing with an X100P is
unacceptable. Does the Audiocodes do better?
2) Line signalling. I'm using Kewlstart with the
Please check cvs. See into libr2.
The README file says that only China and Argentina are available (I'm a
lucky guy, I'm in Argentina :) ).
Regards,
Gus
- Original Message -
From: Musaluke AK [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 01, 2003 12:36 PM
Subject:
Just the sort of newbie question we all hate ;-)
I'm a bit stuck with MOH. I think all is done right and I've read
everyhing I can find, but whenever * tries to do MOH, all that happens
is
'-z: No such file or directory'
Yes, I am on redHat. Yes I have installed real mpg123. Yes, it does
Please send your musiconhold.conf
Or try this:
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
;random = quietmp3:/var/lib/asterisk/mohmp3,-z
- Original Message -
From: Toby Seaman [EMAIL PROTECTED]
To:
Hi,
Anybody seen this error?
Getting an odd error on the console when I place a call from console to
a SIP station. As soon as the station answers, here is the error.
SIP/172.16.10.24-527b answered OSS/dsp
WARNING[1298960704]: File chan_oss.c, Line 679 (oss_indicate): Don't
know how to display
Do we have any Visual Basic or .NET programmers out there? The reason I
say that is this. I would suggest that a program or application be put
together with Visual basic or .NET that will generate the config files for
you by translating what you want to do in telephony terms/jargon to
asterisk
Ernest,
There is a beta load that you can get from the Audiocodes dealer which
is working for us.
We are using their 4-port MP-104 SIP gateway and the only problems we
have with it
are:
1. Outgoing calls go out to the lines in a round-robin fashion. You can
put any number of the
lines in
[EMAIL PROTECTED] wrote:
Do we have any Visual Basic or .NET programmers out there? The reason I
say that is this. I would suggest that a program or application be put
together with Visual basic or .NET that will generate the config files for
you by translating what you want to do in telephony
On Wed, 2003-10-01 at 15:38, [EMAIL PROTECTED] wrote:
Do we have any Visual Basic or .NET programmers out there? The reason I
say that is this. I would suggest that a program or application be put
No reason to put it in a dead end language like VB. At least you might
be able to use something in
Can anyone confirm that the SIP updates in CVS have fixed the channel
leakage and the codec negotiation problem that was happening a few days
ago?
Thanks
dave
--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by
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