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 time
.
>
> - Original Message -
> From: "Julien" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, August 10, 2003 4:41 PM
> Subject: Re: [Asterisk-Users] Asterisk Newbie ...
>
>
> > With this configuration, the 194
0, 2003 4:10 PM
Subject: Re: [Asterisk-Users] Asterisk Newbie ...
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 _
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 ?
uot;Florian Overkamp" <[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
>
x27;m searching 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= in yo
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 a
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 Mes
he handbok-draft without succes.
>For the moment i don't want to use a card to connect to my phone line ...
>Just 1 ata with 2 phones , and a soft phone with their own voicemail.
>
>Julien.
>
>- Original Message -
>From: "Florian Overkamp" <[EMAIL PR
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
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
This is what you need.
Google allows you to enter a parameter called 'site:'
when you do this it searchs that site only.
The list is archived so you always have it available.
Search at google with the following...
site:lists.digium.com
This will search the archive and you will get lot
1 Transcoding is between codecs. ulaw to
g.729 for example
2 I prefer AMP but unless you install it with
[EMAIL PROTECTED] it could be a pain.
3. You need a clock source for meetme and
other features to work so if you don't have any digium hardware you must use
ztdummy
4. Unless yo
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 lear
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
powerfu
> 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
http://www.ast
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 transcodi
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?
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 woul
Sounds like you really need to understand the basics before you go any
further -- the difference, for example, between FXS and FXO. Anyhow,
see if Digit Networks can support your starter kit -- I know Digium
provides excellent support.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[E
Contexts is the concept you're missing here. What context is the usb phone
in when it starts to dial? This is where the extensions would need to be
defined. Contexts can be a little different for some to understand, you
should read the Asterisk handbook or poke around http://www.voip-info.org to
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 direk
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?
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 b
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 m
hi do u have the sip phones extensions in the extension.conf and are they in
the right context (sip-incoming)???
are the sip phone registering to asterisk?? try stop asterisk and reconnect as
asterisk -vvvc to check see them registering...
From: [EMA
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
allo
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