Hi,
> -Original Message-
> I've been getting a few strange things latly..
>
> Mar 1 14:44:07 NOTICE[1142106560]: chan_sip.c:5585 handle_request:
> Registration from
> '' failed for
> '63.169.60.253'
> -- Registered SIP '2767069017' at 63.169.60.253 port 5060
> expires 120
>
> Thi
While trying to add the ability to record memos into my asterisk system
I ran into a small issue with the Record app. If I specify a silence
detection period the application makes the recording and hangs
up,instead of continuing with the dial-plan as it should. Now if I dont
specify a silence
I coded a dialplan that conditionally forwards a call to my cell phone if
no answer on site.
During a test, I received a call (via Voicepulse IAX) which correctly Dial'ed
out to my cell phone (also via Voicepulse) as expected. Fine - it worked
- except that the voice delay was so extreme (> 1
Just this past Saturday I did a emerge -u
world. Going through the package list, they all seemed fine so I let it go
and update everything. Among the things was mpg123-0.59s-r2.
Shortly after the upgrade, there seem to be
immediate audio problems with users that are canreinvite=no
So
Hi All, I am wondering if anyone has seen this particular problem. We're
running Asterisk 0.7.2 on a Debian Sid system (using the Debian package).
I have a queue set up with 4 Zap channels using a TDM400P card. I have
also tried this with soft phones using iaxComm, and the problem appears to
Does anybody know or have good examples of using all functions in a 7960
(SIP)
/Mike
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Hi,
does anyone has an LDAP based directory for the Cisco 7900 Series? I found
some directorys based on
sql, but an ldap directory would allow syncronisation with evolution,
mozilla, multisync (Palm, OPIE, Mobile phones etc...)
Greez Andreas
have a look at the "ChanIsAvail" application.
good luck,
tad
> I could start out by testing to see if the line is busy using GotoIf, if I
> knew how to test for busy. Alternately, I could ring her primary line for 1
> second and go to voicemail if it is busy. Then I could ring both lines
> toge
Ron -
your best bet will probably to launch an agi script at exten => h,1 that
initiates a new call using either a .call file or the manager interface.
there are several messages detailing how to use both options on the wiki
and in the archives.
good luck,
tad
> --__--__--
>
> Message: 7
> From
On Mon, Mar 01, 2004 at 04:09:51PM -0600, Matt wrote:
>
> Hello,
> I found pebble linux, but asterisk is not packaged with it.
Sorry, bad phrasing on my part. *Debian* has asterisk packaged for it,
so just 'apt-get install asterisk' within your pebble instance, and you
should be up and running wi
Title: Garbled Faxes
My incoming and outgoing faxes are garbled and sometimes get disconnected. I remember reading somewhere that I should use u-law for faxes, but I don’t know how to do that. The fax is connected to a Digium FXS card and the calls come in or go out on a Digium FXO card.
This is the Vonage is using.
TKS,
Paul
David Liu wrote:
Hi John,
Is this what you are getting?
Mar 2 10:28:47 WARNING[245776]: chan_sip.c:4978 handle_response: Got 200 OK
on REGISTER that isn't a register
This is what I got too and I have been puzzled. I have tried all sorts of
thing by varying the register message with FWD and it
Hi John,
Is this what you are getting?
Mar 2 10:28:47 WARNING[245776]: chan_sip.c:4978 handle_response: Got 200 OK
on REGISTER that isn't a register
This is what I got too and I have been puzzled. I have tried all sorts of
thing by varying the register message with FWD and it is still appearing
So, sorry I have general question , h.323 dont work on FreeBSD + asterisk
???,,, I need converter h.323 <> sip and codec converter for h.323.
I use FreeBSD 5.2.
Thanks all,
Serge.
- Original Message -
From: "NetOne Administrator" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday,
Hi!
> What is the relationship between when CDR recording occurs and the
> hangup extension is executed. Normally CDR happens before the h
> extension is executed.
In short: Do not rely on h for CDR purposes.
> I use the h extension to clean up for routines, but sometimes it gets
> called to
> On Mar 2, 2004, at 4:34 AM, [EMAIL PROTECTED]
> wrote:
> > In fact, I'm confident enough of this prediction that I'll add $100 to
> > the bounty if the conditions are met within one year of today (i.e.
> > if you prove me wrong).
>
> I'll add $100 with the same conditions. Please prove us wrong.
Hi.
I have a problem, my X100P detects very well the busy tone (remote hang)
when I call to somebody outside throw an X100P card. But when the person
outside callme and he hang, the X100P doesnt detect the remote hang, and
the connection remains open.
I think it happens because the cadences in bo
Sorry, I'm new here, and find some information in Archives 122 MB without
search very difficulty :(
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 8:32 PM
Subject: Re: [Asterisk-Users] Asterisk on Feebsd , pls. HELP !
I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from
Covad (in the US Southwest) and I have sustained 4 calls without a
problem. I prefer to use GSM over G.711to squeeze it down, but that is
my choice. I don't feel that call quality is substandard.
Michael
On Tue, 2 Mar 2004 00
Edit the top level * Makefile to enable this:
OLD_DSP_ROUTINES
then rebuild and reinstall *
Iain
--On Monday, March 1, 2004 7:09 pm -0300 listas iPfone
<[EMAIL PROTECTED]> wrote:
Hi!
Every time i make or receive a call with my x100p i receive that notice:
NOTICE[1234379840]: chan_zap.c:36
On Mar 2, 2004, at 4:34 AM, [EMAIL PROTECTED]
wrote:
In fact, I'm confident enough of this prediction that I'll add $100 to
the bounty if the conditions are met within one year of today (i.e.
if you prove me wrong).
I'll add $100 with the same conditions. Please prove us wrong.
-TJ
smime.p7s
D
Hi,
I've got a question or two about SIP calling channels. As I understand,
there is no facility for Asterisk to make outbound calls as if it were a SIP
proxy.
As I understand it, it is not possible to add an extention that simply
states "if no match so far, try SIP/" (from what
http://www.voip-i
On Mon, Mar 01, 2004 at 08:10:11PM -, Senad Jordanovic wrote:
> Steve Kennedy wrote:
> > On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote:
> >> You could port your numbers to a licenced telco... Install SDSL (or
> >> even ADSL if you have a lot of faith in your current provider
In our last exciting episode, NetOne Administrator ([EMAIL PROTECTED]) said:
> FreeBSD asterisk port is *NUTS*
> Don't use it!
Huh? My port install of 0.7.2 works well. It does core every so often, but
even IAX2 works (somehow). I have a functioning peer connection with
VoicePulse.
> Asterisk com
Many US based carriers will override your callerid if the number is not
a NANPA complient number. i.e. NXXNXX
On Mon, 2004-03-01 at 18:10, wrote:
> Matt,
>
> This is what I'm looking for. Our current PBX system is a supertrunk
> (channel bank) that doesn't have Caller ID. When we switched
On Sun, 29 Feb 2004, Michael Rowley wrote:
> Does anyone have any information on the zaptel driver under freeBSD? I
> know that there has been a 1200$ bounty posted, but wasn't sure if
> anyone with any talent has taken up the project. (I don't really have
> any talent... :| )
>
We have peopl
How feasable is it to get the Monitor app to combine the channels in
pseudo-real-time and have the resulting audio stream out a soundcard?
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Matt,
This is what I'm looking for. Our current PBX system is a supertrunk
(channel bank) that doesn't have Caller ID. When we switched to VoIP, it
shows 000-000- and we don't want that. So adding a DID solve this
issue. I will go and buy it.
Thank you very much.
-Tri.
- Original Me
:-)
The company I'm doing that for is a telemarketing company. By law they have to
send caller id with every call.
Before we did this, there was no caller id sent (i.e. it was blocked)
Soto block it you have to do nothing...
Matt
- Original Message -
From: "" <[EMAIL PROTECTED]>
To
The CGI runs in a separate process, with special permissions.
How is this accomplished? By suEXEC?
You could also setup asterisk to save the vm files as another user.
Michael
This would be the best option IMHO. What is required to do this?
Regardless of whether I use vmail.cgi or a php, using ch
Hi Matt,
Thank you for the info.
Have you test with the SetCallerID(your_number_here) to something likes
SetCallerID(Unavailable ID) with your callerid? I don't actually want to
show my caller ID. Please let me know if you can do that or there is an
option to block caller ID from their website
We have an Audiocodes MP-108 that keeps dropping connections to voicemail
after exactly ten seconds. All other calls are normal, and voicemail works
fine from SIP devices other than the gateway. The reason given for dropping
these calls is "RTP Connection Broken." I suspect that the gateway is
I currently run a multitech multivoip MVP200 over a dialup, simply have
it dlink di804 with an analogue 33k modem hanging off it.
The dialup is through a Satellite circuit of marginal standard, yet the
voip circuit works with both voice circuits in use flawlessly for over
12months with average pin
We had the same problem...
we were trying to set caller id to 6434701641
(64 for NZ 3 for Dunedin rest for our number)
it didn't work...
However, we purchased a DID line for voicepulse and set the cid to that and it
worked.
Maybe because the number was a US number and owned by voicepulse?
Matt
I couldn't get it to work so I switched one box to Linux. :-( Oops, I
better not show that sentiment on this list. :-)
Thanks,
Darren Wiebe
NetOne Administrator wrote:
FreeBSD asterisk port is *NUTS*
Don't use it!
Asterisk compiles just fine on BSD, if you are using 4.x-RELEASE, and
not using
Yes, the plan is to upgrade the wireless bridges to handle QoS.
Mike Machado wrote:
Would like wireless link contain only voice traffic? If not, it would
probably be a good idea to put some sort of minimum bandwidth guarantee
and prioritization.
I have * running over wireless with such bandwidth
On that note, I had been getting warnings about receiving a "200
Registration OK from x.x.x.x - Not a register) or something along those lines.
I tracked it down (finally) to FWD. When I removed the Register statements
for two FWD accounts, those messages went away. I tried adding only one of
I've been getting a few strange things latly..
Mar 1 14:44:07 NOTICE[1142106560]: chan_sip.c:5585 handle_request:
Registration from '' failed for
'63.169.60.253'
-- Registered SIP '2767069017' at 63.169.60.253 port 5060 expires 120
This is a Cisco ATA running 3.1, I've got several others, bu
That is accurate, there are only a couple IPDC softswitches on the market,
you can contact me off line if you would like to talk about this, i am all
too familiar and work with it daily.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Machado
Sent: Monda
Hello,
I found pebble linux, but asterisk is not packaged with it.
- Original Message -
From: "Tim Sailer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 3:11 PM
Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage
> On Mon, Mar 01, 2004 at 10
Hi!
Every time i make or receive a call with my x100p i
receive that notice:
NOTICE[1234379840]: chan_zap.c:3640 zt_read: Fax
detected, but no fax extension
Maybe that is problem with brazilian
lines?
How can i stop it?
Miklos
iPFONE Telefonia IPRua
Caio Graco 735 São Paulo SP iP
FreeBSD asterisk port is *NUTS*
Don't use it!
Asterisk compiles just fine on BSD, if you are using 4.x-RELEASE, and
not using chan_h323, chan_oss, zaptel & libpri.
Darren Wiebe wrote:
Sorry to just come on line now. Have you tried the FreeBSD port?
net/asterisk is the place to look. It alway
Would like wireless link contain only voice traffic? If not, it would
probably be a good idea to put some sort of minimum bandwidth guarantee
and prioritization.
I have * running over wireless with such bandwidth management in place
and it works fine, but not near the volume I would expect with 3
Jason, hi
Don't waste your time on old technology.
This was done on the old komodo phone , sold by net2phone
as a yap jack and there are some ipphones (VIC phone) comes
to mind with an analogue modem built in.
We even have adsl here in parts of Africa, (not that its
got any bandwidth throughput)
I believe that lucent tnts idea of ss7 gateway is using IPDC to talk to
a box that has the ss7 connection. I know of no open source IPDC
implementation.
On Mon, 2004-03-01 at 13:38, Michael Baird wrote:
> We are looking into turning up SS7, my Lucent TNT's support it by using
> a "SS7 Gateway", ar
We are looking into turning up SS7, my Lucent TNT's support it by using
a "SS7 Gateway", are there any open source products that will serve this
purpose. I've looked at openss7.org a little bit, it looks kind of
stagnant, and it doesn't appear asterisk has any of the functionality
I'm looking for.
Hi Mark,
its sets the context in zapata.conf.. put a context= somewhere
above where you define the channels
then in your extensions.conf, create the context if you've not already, start
with a exten=> _.,1,Dial(blah...) then modify from that to suit your setup
Steve
On Mon, 1 Mar 2004, Mark M
Hello!
I am considering deploying asterisk for use in 3 hotels (run as one
company) in downtown Reykjavik, with a total of 100 rooms. The hotels
are connected with a wireless link (802.11) since they are quite close
to each other, the link has been very stable, about 2 hours downtime for
the
Mark,
Zaptel is where it is told to go.
I have mine set to incoming, and a context of incoming in my
extensions.conf.
My configs are here http://www.codepipe.com/id25.htm without any UID's or
PWD's.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Beh
On Mon, Mar 01, 2004 at 10:01:48AM -0600, Matt wrote:
> Hello All,
> I was wondering if anyone is successfully running asterisk on a system
> with solid state storage, such as a compact flash card? I'm looking for some
> pointers on doing this.
Look for pebble linux, based on Debian, which has
[EMAIL PROTECTED] wrote:
Message: 10
Date: Mon, 01 Mar 2004 10:02:54 -0500
From: John Fraizer <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CVS login
Reply-To: [EMAIL PROTECTED]
Glenn Dalgliesh wrote:
I seem to be having trouble with cvs login. anyone having similar pro
Hey...I'm sure that this is something very simple that I'm missing (I've
had * just about 2 weeks or so now and just finally got my x100p card
in). Here's my simple setup.
VoIP Network--*--PSTN via X100P
I can make outbound calls from my VoIP network just fine. However, when
trying to dial in
Thanks Andrew. But it doesn't work either.
I believe that this must be done at the termination end point (VoicePulse
Asterisk Server) since the outbound is going to VoicePulse PSTN line
(PRI/T1). They must set that option on their end.
However, technical support from VoicePulse said that they a
Sorry for the lack of formating in previous email
I have been doing the following and it seems to work fine
# cd /usr/src
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login -the password is anoncvs.
# cvs checkout zaptel libpri
# cvs checkout -rv1-0_stable asterisk
# cd zap
On Mon, 2004-03-01 at 14:30, Andrew Thompson wrote:
> Wim Venneman wrote:
> > Maybe this??: http://www.grandstream.com/y-product.htm
>
> That link didn't bring up anything special... Just the ATA286 and
> Budgetone 102. Is there something I missed?
>
VoIP over modem isn't very good, and probably
Hello, I'm trying to configure our Inter-tel AXXESS (R4.4) system to connect
to an Asterisk system (0.7.2 for now) to initially be a conference bridge,
and perhaps later start hanging VoIP phones off of it so that we can
gradually phase out the AXXESS system (we don't have the budget to do a
forkli
Wim Venneman wrote:
> Maybe this??: http://www.grandstream.com/y-product.htm
That link didn't bring up anything special... Just the ATA286 and
Budgetone 102. Is there something I missed?
-
Andrew Thompson
http://aktzero.com/
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Asterisk-Users ma
Question is: how can I connect Asterisk to another pxb using a BRI or PRI
port ?
I would like to have Asterisk simulate an incoming telco line, providing an
"IP upgrade" to the old pbx, is this possible ?
If yes, what hardware cards you will raccomend for E1 and quad BRI ?
I understand that my
wrote:
> Hi everyone,
>
> Is there anyone know how to block callerid with VoicePulse Connect
> outbound termination? It's showing up 000-000- when I call out.
> Please show me the trick if you know how.
>
> Thanks.
>
> -Tri.
If you'll do a SetCallerID(your_number_here) you can set it t
Maybe this??: http://www.grandstream.com/y-product.htm
- Original Message -
From: "Jason Miller" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 10:53 PM
Subject: [Asterisk-Users] Dial up adapter
I was wondering if anyone has used an adapter to dial up to a loca
Hi all,
I am connecting two PBX using two * and IAX2. There is one E1 connected
to each *. I receive a call from PBX-A and dial to *-B / Zap-g1 (PBX-B).
If the destination is busy or ring until I put on hook, *-A returns
normal Call Clear and the PBX-A (attached to *-A) bills the call.
PBX-A <
Steve Kennedy wrote:
> On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote:
>
>> You could port your numbers to a licenced telco... Install SDSL (or
>> even ADSL if you have a lot of faith in your current provider) and
>> get all lines working throught SDSL.. :) You prabably should ke
Il lun, 2004-03-01 alle 18:14, Andrew Kohlsmith ha scritto:
> Has anyone looked at using busybox and uClibc with asterisk? Those two (and
> agressively stripping everything) were the biggest things in making Linux
> tiny. That and eliminating static binaries whereever possible.
I've run a uCli
On Monday 01 March 2004 13:59, Jim Rosenberg wrote:
> If someone leaves a voicemail message and gets booted out of
> voicemail because the message is too long, I would like different
> behavior than if the # key is pressed. Anyone have a snippet of
> extensions.conf that shows how to do this? How d
I was wondering if anyone has used an adapter to dial up to a local internet service
then used the VOIP phone instead of needing a computer. If so what product do you
suggest? An idea of what I am looking for,
configure the device which has a analog port to dial said ISP and authenticate
Has an
If someone leaves a voicemail message and gets booted out of voicemail
because the message is too long, I would like different behavior than
if the # key is pressed. Anyone have a snippet of extensions.conf that
shows how to do this? How do I do a generic gotoif based on the result
code of an appli
Hi everyone,
Is there anyone know how to block callerid with
VoicePulse Connect outbound termination? It's showing up 000-000- when
I call out. Please show me the trick if you know how.
Thanks.
-Tri.
Q.462 to Q.478 cover R2. You need to know how it has been modified in
each country too, as nobody quite implements the ITU specs.
In some weeks we will buy 405 or 410 bords if it works with Brazilian
ISDN-PRI,
also we will stop using some E1 trunks..
if anyone is able to port R2 MFC-5C signal
On Monday 01 March 2004 13:02, William Waites wrote:
> On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote:
> > server dont have any sound device ( I think:) )
> > Why noone make normal Makefile and FAQ for FreeBSD Asterisk.. als
> > many for Linux.
>
> I don't know why the Asterisk crowd is res
Sorry to just come on line now. Have you tried the FreeBSD port?
net/asterisk is the place to look. It always dumps core on me but you
may have better luck.
Darren Wiebe
[EMAIL PROTECTED]
William Waites wrote:
On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote:
Thanks William, it's get
On Mon, Mar 01, 2004 at 11:37:59AM +0100, Serge wrote:
> Thanks William, it's get.
> but new problem:
> server dont have any sound device ( I think:) )
> Why noone make normal Makefile and FAQ for FreeBSD Asterisk.. als many for
> Linux.
I don't know why the Asterisk crowd is resistant to
usin
On Mon, Mar 01, 2004 at 06:23:08PM -, Senad Jordanovic wrote:
> You could port your numbers to a licenced telco... Install SDSL (or even
> ADSL if you have a lot of faith in your current provider) and get all
> lines working throught SDSL.. :) You prabably should keep one BT line
> for fax and
On Mon, 01 Mar 2004 07:34:33 -1000, Jean-Denis Girard <[EMAIL PROTECTED]>
wrote:
>Download link at top of page is broken for linux
>iaxcomm-lin-20040228.zip should be iaxcomm-lin-20040228.tar
Thanks. Fixed
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On Mon, Mar 01, 2004 at 05:32:34PM +, WipeOut wrote:
>
> Currently connecting more than 3 analog lines to asterisk can be
> problematic unless you get hold of a channelbank (not that availible in
> the UK)..
>
Of course there is a 12 port configurable FXS/FXO
blade from VoiceTronix
/w
Well, I finally found my own answer, so I'm posting to share.
In php, $argv is an array of the command line variables passed to the
routine. So, if in extension.conf you do a :
exten => AGI(myphpcode.php,${EXTEN})
$argv[1] will have the value of EXTEN. Note that it's $argv[1] and not
$ar
WipeOut wrote:
> Angel Gabriel wrote:
>
>> I have 5 BT phone lines coming into my office. We use four for
>> international calls, and one for local/mobile calls. We have just
>> obtained another call carrier, and now we would like to be able to
>> make calls from any phone to any carrier, without
http://ftp.gwdg.de/pub/suse/i386/9.0/boot/rescue
best regards
Klaus
Am Mo, 2004-03-01 um 18.38 schrieb Matt:
> Hello Klaus,
> Is it possible for me to download an image of the os or can you point me to
> the rescue disk that you used?
> - Original Message -
> From: "Klaus-Peter Junghann
Kelly Murphy wrote:
Follow all the instructions on
http://www.asterisk.org/index.php?menu=download You still need to
checkout libpri and zaptel. If you want more information checkout
http://www.voip-info.org This is the main repository if information on
Asterisk.
>> No you don't need to checko
- Original Message -
From: "NetOne Administrator" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 5:25 PM
Subject: Re: [Asterisk-Users] Small office requirements - Can this be done?
>
> Angel Gabriel wrote:
>
> > I have 5 BT phone lines coming into my office. We
I am using ztdummy to get timing as I do not use digium-hardware, which
is part of zaptel. So I guess I have to use both zaptel and libpri??
Correct me if wrong. So my question is where to get the stable set of
libpri and zaptel? It is useful, if asterisk.org -> Download be updated
with little more
On Sunday 29 February 2004 20:13, Michael Rowley wrote:
> Does anyone have any information on the zaptel driver under
> freeBSD? I know that there has been a 1200$ bounty posted, but
> wasn't sure if anyone with any talent has taken up the project. (I
> don't really have any talent... :| )
If t
Hello Andrew,
I'm looking on some detailed information on how to build a working
system that small. I can't seem to find a whole lot, on what asterisk
requires to run.
Any info you could give me would be great. Feel free to email me directly.
Thanks
-Matt
- Original Message -
From: "
Michael Van Donselaar a écrit :
There are new iaxComm binaries for Windows, Linux and Mac OSX posted at
http://iaxclient.sourceforge.net/iaxcomm/index.html
These binaries also have the recent library change that allows client to client
connections to be handed off correctly.
Recent changes inclu
Does anyone have a working AGI script to detect DTMF within a Meet-Me?
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Angel Gabriel said:
> I have 5 BT phone lines coming into my office. We use four for
> international calls, and one for local/mobile calls. We have just obtained
> another call carrier, and now we would like to be able to make calls from
> any phone to any carrier, without having to remember what d
Hello Klaus,
Is it possible for me to download an image of the os or can you point me to
the rescue disk that you used?
- Original Message -
From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 11:54 AM
Subject: Re: [Asterisk-Users] Tiny in
I have been doing the following and it seems to work fine# cd /usr/src#
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot# cvs login -
the password is anoncvs.# cvs checkout zaptel libpri# cvs checkout -r
v1-0_stable asterisk This will create just the asterisk directory.
Comp
Angel Gabriel wrote:
I have 5 BT phone lines coming into my office. We use four for
international calls, and one for local/mobile calls. We have just
obtained another call carrier, and now we would like to be able to
make calls from any phone to any carrier, without having to remember
what det
At 08:51 AM 3/1/2004, you wrote:
I have 5
BT phone lines coming into my office. We use four for international
calls, and one for local/mobile calls. We have just obtained another call
carrier, and now we would like to be able to make calls from any phone to
any carrier, without having to remember
Follow all the instructions on
http://www.asterisk.org/index.php?menu=download You still need to
checkout libpri and zaptel. If you want more information checkout
http://www.voip-info.org This is the main repository if information on
Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
Ali Mughrabi wrote:
Hi ,
I need to execute a query when a user hangs up the agi application ,
I’ve tried monitoring some return values of AGI commands
Still doesn’t work .
Any ideas ?
Thanx
Ali Mughrabi
You will need to put another agi with you cleanup script onto the 'h'
extension.. If yo
Angel Gabriel wrote:
I have 5 BT phone lines coming into my office. We use four for
international calls, and one for local/mobile calls. We have just
obtained another call carrier, and now we would like to be able to
make calls from any phone to any carrier, without having to remember
what det
If you are not using Digium hardware, then you don't need libpri and zaptel.
Asterisk WILL build on its own.
SamW wrote:
I want to build a stable asterisk to run, if some one can guide through
how to compile will be useful. Currently available documentation do not
show any good information about a
None of this is mine, but it's useful all same :)
http://www.xs4all.nl/~hreuver/net4501-try1.html
http://www.antlinux.com/staticwiki/LinuxOnSoekris.html
At 08:49 AM 3/1/2004, you wrote:
At 08:33 AM 3/1/2004, you wrote:
>I'm curious what distro of linux you used. I also can't seem to find a
>listin
> We are working on trying to build this in 100MB... or less. Stay tuned.
I've created fully functional Linux installs in under 16MB (9.8MB IIRC, but
the entire thing, including the configuration partition was stored on a
16MB CF card) -- I know that Linux can fit on a floppy but my builds
incl
Thanks Wes, I just tried it but it does not seem to make any difference.
Darren Wiebe
Wes Marderness wrote:
My server is running fine now. I have to 'cd /tmp' then
'/usr/sbin/asterisk -gc' or I receive error messages. It is very strange
but it works.
Wes
-Original Message-
From: [EMAI
Thanks for the info, that would be great if you could find the intructions.
-Matt
- Original Message -
From: "Ernest W. Lessenger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 10:49 AM
Subject: Re: [Asterisk-Users] Tiny install with Solid State Storage
> At 08
Matt wrote:
Hello John,
I saw the wiki page on trustix, it said 296 megabytes, still a little
big. I'm downloading trustix now to check it out though.
Thanks
-Matt
Trustix can be made a lot smaller by dumping the kernel source after you
have compiled Asterisk or by building RPM's an only in
Hi,
I am running * on a modified SuSE 9.0 rescue system. Total system
including sshd, *, MOH and * prompts is 32 MB zipped. It expands
to 52 MB on a 64 MB RAM disk. I boot it from a compact flash disk.
The system is a 600 mhz transmeta crusoe with only 110mb ram. It is
powerful enough to drive a
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