Hello Everyone, I am new to Asterisk and telecommunications, and I am
lucky to have found this mailing list. Is there a simple guild for
setting-up a simple GSM Asterisk system, or better yet can someone
please mentor me through the process?
Art
--
On Thu, 26 May 2011, Art wrote:
Hello Everyone, I am new to Asterisk and telecommunications, and I am
lucky to have found this mailing list. Is there a simple guild for
setting-up a simple GSM Asterisk system, or better yet can someone
please mentor me through the process?
My suggestion
I would like to the know following:
1. What is the latest greatest asterisk verision? and how to get it.
2. can i run into with linx FC4 and kernel 2.6
3. how can i contribute to development of IM and Presence work on asterisk.
Thanks
roswel
___
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of roswel
ajfSent: Tuesday, January 24, 2006 10:56 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] asterisk
newbie
I would like to the know following:
1. What is the latest greatest asterisk verision
Hiii ; actually you are not allowing any codecs in the sip.conf neither alaw nor ulaw
so try this to all phones in sip.conf or put it in the general context (allow=all)
[2011]
type=friend
username=2011
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
Hi list, i'm an asterisk newbie and i've to setup a net with an
asterisk server and several ip phones linked on the net.
i hope my questions are IT ans if you have some link for solving those
problems please mail me.
i've wrote the sip.conf in this way:
[2011]
type=friend
username=2011
secret=1945
: [EMAIL PROTECTED] on behalf of Sistemista WebSolvingJaa
Sent: Fri 7/1/2005 6:43 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk newbie and phones which don't want
tocomunicate
Hi list, i'm an asterisk newbie and i've to setup a net with an
asterisk server and several ip
You have to put entries in sip.conf
Race the Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michele
O-Zone Pinassi
Sent: Friday, May 13, 2005 6:47 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk newbie
I've
I've just installed Astrisk with AMP. All work well but one thing is not
clear. I wanna add users to allow calls between SIP phones. I've added
extension but seems not to be enought.
How i can add SIP users and allow calls between they ?
Thanks ! Oz
--
O-Zone ! No (C) 2005
Hello
all
I have been learning
* from almost 1 month now. It looks really powerfull. I have some problem trying
to find previous post, or solutions to common problems, advice to newbies etc in
this mailing list. There is noa forum-like tool to search thru the
posts by keyworks for example.
On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks
really powerfull. I
have some problem trying to find previous post, or
solutions to common
problems, advice to newbies etc in this mailing list.
There is
:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk
Newbie
Hello
all
I have been learning
* from almost 1 month now. It looks really powerfull. I have some problem trying
to find previous post, or solutions to common problems, advice to newbies etc in
this mailing list. There is noa
-
From:
Fabian Borot
To: asterisk-users@lists.digium.com
Sent: Tuesday, March 15, 2005 10:56
AM
Subject: [Asterisk-Users] Asterisk
Newbie
Hello
all
I have been
learning * from almost 1 month now. It looks really powerfull. I have some
problem trying to find
To search the list archives use this in Google:
site:digium.com search-terms
-Original Message-
From: Fabian Borot [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 15, 2005 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Newbie
Hello all
I have been
Or if google is too complex, http://asterisk.keystreams.com
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Robert Webb wrote:
On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks really
powerfull.
forum-like tool to search thru the posts by keyworks for example.
You can use google by specifying site:lists.digium.com before or after the words
Most if not all of your questions are answered on the wiki (which does
not seem to be responding as I write this) and at sites like
Fabian,
Searching is a good start, but here are the answers to your questions anyway:
1- Transcoding: is this when you go from g711 to g729 for
example? Or when you go from SIP to IAx?
Transcoding is converting audio data between codecs, like G711 -
G729. I wouldn't call SIP to IAX transcoding
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This is where the
Voip WIKI tells me to go for
On Sat, 11 Sep 2004, John Stegenga wrote:
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This
On Sat, 11 Sep 2004 10:01:27 -0400, John Stegenga [EMAIL PROTECTED] wrote:
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
Some people may have a filter in their inbox that has newbie in it
going directly to trash. Just kidding, it's been a
John Stegenga wrote:
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This is where the
Voip WIKI tells
Hi everyone.
I'm a bit of a Linux newbie, but I've been doing tech stuff for ages.
I'm also brand new to *.
I've been reading the Voip.org wiki, and perusing the list archives for a
while since I've been asked to investigate using IP telephone / soft phones
for a call-center type scenario. People
Hi John,
I'm also new to *, but if you want to set up a callcenter, with 40
people calling the same number at the same time, you probalbly will need
a T-1 or E1 line wich AFAIK handles at least 30-calls.
You then need at least one Digium E1/T1 card to get the calls into * and
other cards to
Hi
I've been playing around with asterisk for a while now at home, just trying
to understand a bit of the technology and seeing what I could get up and
running. Here's where I am at:
I bought myself an X100P card and got an asterisk server up and running on a
gentoo linux distro. I got two
hi,
I got a digit networks x100p card and instaled asterisk. everything went
fine and upon calling the phone asterisk issues a notification. Now i plan
to turn it into an ivr and modified extensions.conf to first record some
messages , problem is
1-)I am not able to understand how extensions
:[EMAIL PROTECTED] On Behalf Of digvijay
singh
Sent: Friday, June 11, 2004 8:16 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk newbie help !!
hi,
I got a digit networks x100p card and instaled asterisk. everything went
fine and upon calling the phone asterisk issues a notification
] On Behalf Of digvijay singh
Sent: Friday, June 11, 2004 9:16 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk newbie help !!
hi,
I got a digit networks x100p card and instaled asterisk. everything went
fine and upon calling the phone asterisk issues a notification. Now i plan
to turn
On Mon, 2003-08-11 at 11:28, Julien wrote:
Just a last question, if i configure G723 in my ATA, i can't call the
voicemail for exemple. I've seen that messages were in GSM format. Is there
a way to be able to acces to the voice mail in G723 (for remote users) and
in G711 for local users ?
In
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 10, 2003 3:47 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...
At 15:13 10-8-2003 +0200, you wrote:
If i want to call the sjphone from the ata or call the ata from de
sjphone
everything is ok.
My problem is ,that i can't call
Hi ;)
I'm a french newbie and i installed asterisk 1 day ago.
I've got an ATA186 and a computer with Sjphone installed.
If i want to call the sjphone from the ata or call the ata from de sjphone
everything is ok.
My problem is ,that i can't call the voicemail or any other phone number
..as 600
and thanks for your help.
Julien.
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 10, 2003 4:49 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...
Julien,
try adding defaultip=ip of phones in your sip.conf for each phone
Fabia,
The only numbers you should be able to dial from that config are
1945
1943
2999
and nothing else...
The entry under bogon-calls (isn't it bogus calls?) should read
exten = s,1,Congestion
rather that using the _. ...
HTH
Andy
*** REPLY SEPARATOR ***
On 10/08/2003
: Re: [Asterisk-Users] Asterisk Newbie ...
With this configuration, the 1943, 1945 are available , it's ok
but the 2999 is not available... In sjphone 404 error, on the ata busy
tone
...
Julien.
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent
PROTECTED]
Sent: Sunday, August 10, 2003 4:41 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...
With this configuration, the 1943, 1945 are available , it's ok
but the 2999 is not available... In sjphone 404 error, on the ata busy
tone
...
Julien.
- Original Message
At 15:13 10-8-2003 +0200, you wrote:
If i want to call the sjphone from the ata or call the ata from de sjphone
everything is ok.
My problem is ,that i can't call the voicemail or any other phone number
..as 600 for exemple from the ata or the jphone.
I don't know why but i looked after a long
35 matches
Mail list logo
