First of all thanks for the patch it works great,
but i think it breaks the distinctive ringing,
I have 2 incoming numbers in one x100p in contexts home1 and home2 but
'default' is always chosen has anyone else seen this?
if you need any more info just ask
Robb
Tony Hoyle wrote:
David J Carter wr
Aaron J. Angel [EMAIL PROTECTED] wrote:
> > Did you know that by clicking on "REPLY" instead of creating
> > a new message/thread many people that could have helped you
> > won't see your message? Those are the people that don't care
> > about VoicePulse and so just delete messages about Voicepuls
Hi,
Is anybody using "contexts" successfuly with "mysql-vm-routines"?
Everything works well except for the fact that the voicemails are not
left in their respective context(the one defined per mailbox in the
mysql users table). They are all dropped in the main directory. I we
switch back to de
I am trying to integrate asterisk with a 3rd party PBX for voicemail
(Mitel). For the most part, things are working well. We only have one
main issue left: hangup detection. The connection between the two is:
* w/T100P -> Zhone channel Bank FXO port -> Mitel ONS (station) port
(Yes, overkill, but
On Thursday 27 May 2004 18:09, Mike Stupak wrote:
> I seem to have gotten asterisk installed and running w/ the sample configs,
> I have a TDM-400P (w/ 4 FXO modules). Im using fedora FC2.
>
> As part of testing/learning I set up Kphone on the same box. Tweeked the
> sip.conf (per instructions on
Hi All,
I'm new to asterisk, and so far have yet to get past running the server up
on a test PC.
I have 2 Cisco 7960 phones to play with, both upgraded to the latest SIP
image (7.1)
I'd like to do 2 things, and hope that someone can point me to some simple
documentation, example configs or other
This one is making me crazy. I have a T1 connection from * to an adit
channel bank. When there is voicemail, there is not stutter for message
waiting indicator. Here is zapata.conf.
Thanks!
[channels]
language=en
signalling=fxo_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
call
Hello all,
I spent a little time in the wilderness and when I came back to
civilization, I found out that new Libpri, Zaptel and Asterisk archives
were available on ftp.digium.com. As a result, I've updated my Asterisk
RPMS for RedHat and made them available for public consumption.
On Thursday 27 May 2004 12:51, Eric Wieling wrote:
> Did you know that by clicking on "REPLY" instead of creating
> a new message/thread many people that could have helped you
> won't see your message? Those are the people that don't care
> about VoicePulse and so just delete messages about Vo
Michael George wrote:
> But, this isn't a big deal, we can live without it. I just
> thought there might be a way. If I could do a
> Backtround(Playtone()), that would do what I want...
There's no need for that. The playtone application continues to the next
priority as it plays the tone, and
Hi All,
Just thought I’d provide a quick
summary on this thread as the problem is now resolved.
I didn’t actually hear what the
cause of the problem was as Digium shelled into my machine last night (Australian
time GMT+10 here :-) and installed the codec and checked the license regi
Shift-pgup is your friend
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Togan
> Muftuoglu
> Sent: 28 May 2004 04:26
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Scroll mode in cli
>
>
>
> Hi,
>
> Is it possible to have the Asterisk CLI to s
This may sound stupid but could you not off load the 4 pri max limit off to
a voice router. I know it's not the cheapest way to go since dsp for a
hdv-2mft (cisco card) can run you a few K but would make it so your limited
by the Voice router limit so you get higher number of pri to a * box? If s
Hi ""noobs"" and others(like me).
I've been struggling for hours,days,days,days to get my Asterisk work
with 2x HFC-S BRI cards.
One card for use in NT-mode for connecting my ISDN DECT phone,and one card
as gateway to PSTN.
Now it'works,and I just want to share some of the experience with oth
On Thu, 2004-05-27 at 12:11, David H Hickman wrote:
> Is there any reference for the dg-104 telnet(shell)
It is the same as the serial port. Login is same as webserver.
>From the manual.doc
General Setup Commands
nwdbg system reboot
Definition: This command is
On Thu, 27 May 2004, Harry Flink wrote:
> www.cvshome.org is home for CVS but the site is currently down.
Is down due to security issues:
. TA04-147A - CVS Heap Overflow Vulnerability (US-CERT)
http://www.us-cert.gov/cas/techalerts/TA04-147A.html
. Advisory 07/2004 - CVS remote vulnerability
On Thu, 27 May 2004, Mike Stupak wrote:
> However, when I try dialing 1000 I get the following error message:
>
> "Close any program which might be using soundcard and then retry dial".
If you're using the ALSA sound server, I think this might be the problem.
It opens /dev/dsp and doesn't cl
Mike Stupak <[EMAIL PROTECTED]> wrote:
I seem to have gotten asterisk installed and running w/ the sample configs,I have a TDM-400P (w/ 4 FXO modules). Im using fedora FC2.As part of testing/learning I set up Kphone on the same box. Tweeked thesip.conf (per instructions on wicki), and it looks like
Adam Goryachev wrote:
I suppose I could do QoS on outbound, which should improve things
somewhat for the remote caller, but that doesn't help inbound packets.
Does anyone have any comments on what this would mean for VoIP calls
with the above variables?
I think the biggest problem is the jitter
Scott Stingel wrote:
Hi Steve-
Just briefly:
I was mentioning the old days to illustrate what an even low clock rate DSP
can do. More recently (2000-2001), using D/600's we were able to drive a
large number of channels (8-12 E1's) for IVR.
Ah, the D/600 - damned big heat sink, lots of heat, and
I seem to have gotten asterisk installed and running w/ the sample configs,
I have a TDM-400P (w/ 4 FXO modules). Im using fedora FC2.
As part of testing/learning I set up Kphone on the same box. Tweeked the
sip.conf (per instructions on wicki), and it looks like it connected just
fine. However
On Fri, 2004-05-28 at 10:36, Adam Hart wrote:
> cts-au.freshtel.net sorry, it's hosted at comindico in sydney.
Nicer...
I get over 100 pings
min 14ms
avg 34ms
max 234ms
with one packet dropped.
(bad my end, I have bursty traffic for SMTP/POP server)
I suppose I could do QoS on outbound, which sh
On Fri, 2004-05-28 at 05:37, Michael George wrote:
> On May 27, 2004, at 2:01 PM, Rechenberg, Andrew wrote:
> > I believe it's the 'ingnorepat' option that you want. Look at the
> > stock
> > extensions.conf and search for ignorepat.
>
> I've tried ignorepat => 9, but that only seems to work wit
tim panton wrote:
Steve Underwood wrote:
Jason Williams wrote:
At 09:16 27/05/2004 -0500, you wrote:
Maybe the time and effort would be better spent finding out why the
Digium card won't work on the NTL's PRI and either fixing it or
providing the information and testing facility to someone who can.
Adam Goryachev wrote:
On Fri, 2004-05-28 at 09:28, Adam Hart wrote:
If anyone's after Australian IAX termination (or Australians wishing to
call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net
Except I get:
[EMAIL PROTECTED]: ~$ mtr ctsau.freshtel.net
mtr: Unknown host
Perhaps
On Fri, 2004-05-28 at 09:28, Adam Hart wrote:
> If anyone's after Australian IAX termination (or Australians wishing to
> call overseas), try www.freshtel.net - iax server is ctsau.freshtel.net
Except I get:
[EMAIL PROTECTED]: ~$ mtr ctsau.freshtel.net
mtr: Unknown host
Perhaps you could just l
Hi Paul-
I actually said 4 E1's to a chassis (one TE410P) was best I've been able to
do. And yes I do believe that it is the short call profile (large number of
call setups) that contributes to the limit - if you're doing longer calls
then the limit seems to be more related to the number and type
This really only seems practical in a corporate environment however, I would
suggest decoupling the creation of faxable material from the act of
transmitting it.
First use CUPS/Samba to create a 'print to PDF' printer
(http://www.google.ca/search?q=samba+pdf+printer)
Second use an LDAP to look up
David J Carter wrote:
Where would I find cdr-csv?
Usually in /var/log/asterisk
The line looks funny because of the line breaks.
zapata.conf
ukcallerid=yes
callerid=asreceived
signalling=fxs_ks
channel => 1 : BT line
channel => 2 : Telewest line
I also have immediate=yes, but that shouldn't affect a
On Thu, 2004-05-27 at 15:26, Togan Muftuoglu wrote:
> Hi,
>
> Is it possible to have the Asterisk CLI to scroll certain amount of
> lines and wait for, say spacear to be pressed to continue. If so how ?
Why? Don't you know how to configure your scroll buffer so you can get
backwards to where you
They'll be a new version at the end of the day (it's 9:25am now) - The
reason it was like that was to cope with overlap for the firefly network
going to Freshtel. Freshtel will have the Firefly Network and special
version of Firefly (no IAX and SIP) while Virbiage will have a standard
IAX and S
FYI Downgrading to -stable totally fixed the choppy audio on Cisco my 7960
<- * -> IAX setup. Now, when would a fix that goes into stable get into the
current source (HEAD)? And, isn't checking stuff into a stable branch that
doesn't exist elsewhere in the source tree break some rules somewhere?
usedcanon wrote:
Quite interesting, since there version history say 1.4 is the latest. The
one you download is 1.7 and only works with Firefly. I have V1.5 which has
the option to connect to other services.
I am interested to know whats the highest version anyone has that has the
other services opt
Hi Matteo,
Thanks, suddenly makes sense now. I guessed that is the case however was not
sure. Any opinion on what is more/most efficient, using a scripting language
like perl or a compile app in C/pascal.
What I am looking to do is some database access with in the script to rate a
call and set an
Where would I find cdr-csv?
I have looked at all the asterisk directories.
CLI is on all 3 lines I have into the house.
> I have addes the ukcallerid=yes to my zapata.conf, and also got
> callerid=asreceived set.
The line looks funny because of the line breaks.
zapata.conf
ukcallerid=yes
call
brian,
I've tried repeated emails with both direct to jeremy and to company
address over the last two months. No response, but I know the emails
were delivered. I gave up.
Rich
> Nufone does the forwarding for us during an outage... I still don't know
> why you guys don't get emails from them..
Hi
Il gio, 2004-05-27 alle 23:32, usedcanon ha scritto:
> Hi,
> Has anyone done any AGI scripting in pascal. I would appreciate help anyone
> can offer. My understandin on AGI scripting is very flaky, I am assuming
> whatever language is used the application needs to be compile and made
> executab
Hi again!
> i upgraded to the actual CVS head from yesterday (27.5.) but can not get
> incoming SIP calls from my provider (sipgate). If someone calls my
> number, my asterisk responds with the following error:
More details: SIP DEBUG on the called server reveals this:
Found peer 'fwd-outgo
===8<==Original message text===
On May 26, 2004, at 1:32 PM, Stephen Karrington wrote:
> Hello,
>
> We are proud to officially release our worldwide IAX termination
> service.
Your rates page still has no useful information. Can you guys at least
put some rates up somew
Hi all,
I am looking for a software package (free or not), or an
inexpensive hardware device, which can route calls between
an H.323 network and an MGCP-based voice service.
Unfortunately, I believe (based on documentation and other forum
posts -- I have not looked at the code) that Asterisk can
I can second this. Clearpath=v.good. I have one DID and an 800 through
them.
Michael
On Thu, 27 May 2004 10:49:07 -0400, John Fraizer wrote:
>> Steve Totaro wrote:
>>
>>> I went with voicepulse after I emailed NuFone sales twice about paying for
>>> some 800 numbers that were never responded to
Is that in a new rev of software? I have v1.4 and I LOVE it! Best soft
phone I've ever used. Would be a pity if they closed it down to only
their service. Totally wrongheaded in todays world.
Michael
On Thu, 27 May 2004 09:32:13 -0500, brian wrote:
>Just an FYI FireFly no longer works with anyth
Hi Tim,
TP> So it _may_ not be a problem for me as NTL is a patchwork
TP> of smaller telcos, my area (Manchester) may be more up to
TP> date.
TP> Anyone know an easy way to tell what I've got ?
TP> (or will I have to ask NTL -gh)
"Pound to a penny" you have ISDN 85. It's been reported via the
Hi,
Has anyone done any AGI scripting in pascal. I would appreciate help anyone
can offer. My understandin on AGI scripting is very flaky, I am assuming
whatever language is used the application needs to be compile and made
executable. So if I write a script in pascal, I would compile it with
somet
David J Carter wrote:
I have addes the ukcallerid=yes to my zapata.conf, and also got
callerid=asreceived set.
No idea what that option does...
The phones now ring without the screen showing starting simple switch 3-4
times, but alas no callerid on my GS phone.
Check your cdr-csv file to see if ast
> spandsp and HylaFAX will not currently play together. I did some work
> towards giving the FAX modem in spandsp and optional Class 1 AT
> interface, but it is unfinished.
>
> Regards,
> Steve
Okay, not that I know anything about programming, telephony, SpanDSP, or
modems in particular, let's
Hi!
> i upgraded to the actual CVS head from yesterday (27.5.) but can not get
> incoming SIP calls from my provider (sipgate). If someone calls my
> number, my asterisk responds with the following error:
>
> May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request:
> Failed to authe
Title: Message
Hi
All,
I found app_radius
and cdr_radius, but I'm not sure how to use them. They are poorly documented.
There are a few open source billing systems (trabas) out there that use radius
as the cdr method and I'm wondering how this can be integrated with *. Just
adding cdr_radiu
Hi all,
First an introduction. I've been using Asterisk at home for a few months,
but only just joined the list. So far I'm very impressed.
I'm glad I'm not the only one who's been having the problems Julian has.
This is what I get when I try placing a call into my asterisk system. I'm
using voip
> I have no objection to mirrors, but all of the pages are
> dynamically generated and the software the wiki is
> currently running on doesn't provide any easy way to create
> and/or update mirrors.
MySQL replication across the net? Maybe with SSH tunnelling?
Never done it - all my replication stu
> Lets see. Early 90s would be ISA cards in an industrial
> PC chassis full of ISA slots. Heavy IVR means talking
> most of the time. 3k bytes/s per voice for the commonest
> 24K ADPCM mode most people use with Dialogic. So, 12 E1s
> is 360 channels. 3kbytes x 360 = more than the ISA bus can
> hand
Hermann Wecke wrote:
On Thu, 27 May 2004, Tony Mountifield wrote:
No more SIP, No more IAX. It was a damn good IAX client... too bad its crap
now.
Are you sure?
http://www.virbiage.com/firefly/download/ still says the following:
[...]
I just download the latest version (1.7 Build 3532) and they ar
Hi,
Is it possible to have the Asterisk CLI to scroll certain amount of
lines and wait for, say spacear to be pressed to continue. If so how ?
I am using the CVS stable version
Thanks
--
Togan Muftuoglu
___
Asterisk-Users mailing list
[EMAIL PROTECTE
I had been using the HEAD branch, but went back to the STABLE version
just so I could use the drivers that I downloaded from Voicetronix.
Curiously, a majority of my issues were solved when I reversed the
polarity on the incoming line.
But I am now having an error appears indicating that it can'
Quite interesting, since there version history say 1.4 is the latest. The
one you download is 1.7 and only works with Firefly. I have V1.5 which has
the option to connect to other services.
I am interested to know whats the highest version anyone has that has the
other services options.
Umar.
--
We are going to replace a BizFone with Asterisk and one of the features
that I haven't been able to emulate is the PBX hold. We can put a call
into hold with the * key and every 30 sec the phone gets a beep to
remind us it is there and the caller on hold can hit # for other
options.
The close
Steve Underwood wrote:
Jason Williams wrote:
At 09:16 27/05/2004 -0500, you wrote:
Maybe the time and effort would be better spent finding out why the
Digium card won't work on the NTL's PRI and either fixing it or
providing the information and testing facility to someone who can.
NTL's PRI uses
Which termination service are you guys talking about? I didn't see a name in your
message.
Sincerely,
Stephen Karrington
Dreamtime. net Inc.
http://www.diamondcard.us
http://www.dreamtime. net
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
> ignorepat => 9
Ahhh... see that is something I did not grasp the concept of until now -- what
'ignorepat' actually did. Now I know that it defines the pattern of leading
digits received that Asterisk will NOT stop playing dialtone upon receiving.
I wonder if the documentation's been updated
Title: Cisco 7940/60 sip downloads
I know this has been hashed out ad nauseum over time but I am interested in getting the SIP firmware for the 7940/60 and the associated minimal contracts for that software but am just getting the runaround from cisco. The wiki was helpful, and some of the com
On May 27, 2004, at 2:01 PM, Rechenberg, Andrew wrote:
I believe it's the 'ingnorepat' option that you want. Look at the
stock
extensions.conf and search for ignorepat.
I've tried ignorepat => 9, but that only seems to work within a
context. If my context does a Goto after a 9 (into a different
hi
can you send me the 7.1 firmware?
many many thanks for help :-D
i have converted my 7960 two weaks ago with the 6.3 version.
maybe i can help you inconverting - but i´m in hollyday (now and for next
two weeks).
i also have many of ringtones - if you search someone...
regards
thorsten
- Or
> > You deserve a large flaming over not reading for that.
> >
> > Could it be you have not installed CVS..
> >
> > Lazy, non reading person, scatch that, you just rated below person.
> > --
> > Steven Critchfield <[EMAIL PROTECTED]>
> >
I think we should create a PHP page that gives
I believe you will need a "real" phone number to sign up, and they'll
provide 1+ long-distance on that. I have my home phone on zone-ld.
Once you have the account set up, you can choose any number of 800#s (or
transfer them) and have them terminate on your VP incoming line. Or
your cell-phone. O
Hello folks,
i upgraded to the actual CVS head from yesterday (27.5.) but can not get
incoming SIP calls from my provider (sipgate). If someone calls my
number, my asterisk responds with the following error:
May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request:
Failed to authentic
On Thu, 2004-05-27 at 12:56, Steve Totaro wrote:
> Steven,
>
> Sometimes I wonder if you have a big smile on your face when do that. Well
> do you?
Nope. You should have seen the messages from others in IRC. The big
smile came from messages from Brian.
> - Original Message -
> From: "S
===8<==Original message text===
On Thu, 27 May 2004, Chris Sullivan wrote:
> Your rates page still has no useful information. Can you guys at least
> put some rates up somewhere?
They have their rates available at
http://www.diamondcard.us/exec/voip-rep-acc-type?secRel=/se
Hi Steve-
Just briefly:
I was mentioning the old days to illustrate what an even low clock rate DSP
can do. More recently (2000-2001), using D/600's we were able to drive a
large number of channels (8-12 E1's) for IVR.
All I'm trying to do is to illustrate both the beauty and the limitations of
On Wed, 26 May 2004, Steven Sokol wrote:
> I have a client who wants to allow callers to dial a DID which connects
> over a PRI to Asterisk. Asterisk will be analyzing the ANI data from
> each call to that DID and if it recognizes the ANI, it needs to
> effectively return an "Invalid Number" or "
Michael George [EMAIL PROTECTED] wrote:
> The way I have my dialplan configured, an internal extension is routed
> to a different context (with Goto()) on pretty much the first button
> press.
>
> 2 -> internal extensions
> 0 -> operator
> 5 -> VM
> 9 -> outside line
> etc.
>
> So a "201" will g
There is a dummy driver calles ztdummy which can loaded to have those
functionality.
Have a look onto
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
Regards
Julian Pawlowski
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
pesb [EMAIL PROTECTED] wrote:
> I need to implement a SIP Conference Server. I've saw that
> asterisk has an application called meetme. But, it says that "A ZAPTEL
> INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY".
> Is there any other way to implement a conference server without the ne
On Thu, 27 May 2004, John Congdon wrote:
> I know that you can announce the current
> hold time to the caller, and that this hold time
> is based on the box car filter...
>
> Are there any current plans to include
> a management type of feature to export
> actual average, min, and max hold time
With this capi.conf and two AVM controllers (one PCI, une USB) with hacked drivers, we do have
> random problems, when we have many calls, our server hangs and we
must reboot.
Should be up to your hacked drivers, not up to asterisk.
Use active cards (or at least one) to prevent those failures. Ma
Michael George wrote:
Apparently there is no mirror or anything for it? I've been in the
groove for a couple days making great progress, but I need the
application documentation...
You can always use the Google cached pages. I "survived" :-) today
searching at google and reading their cache in
In article <[EMAIL PROTECTED]>, I wrote:
> In article <[EMAIL PROTECTED]>,
> brian <[EMAIL PROTECTED]> wrote:
> > Just an FYI FireFly no longer works with anything but the FireFly network.
> >
> > No more SIP, No more IAX. It was a damn good IAX client... too bad its crap
> > now.
>
> Are you su
In the docs for the zapata.conf file I see a setting called call
waiting, presumably for giving the call-waiting beep to an analog phone
rather than just returning immediately from Dial() with a return value
of 0 and the priority increased by 101.
In short, I would like to:
Rather than catch a
If you don't have CVS, then you probably also don't have the kernel source,
the development tools, etc. What Linux (hopefully) distro are you using?
--Ernest
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Hermann Wecke
> Sent: Thursday, May 27
I'm getting a Got SIP response "400 Bad request" from 172.16.0.105
(which is a Zyxtel 2000w).
It also will not register.
Any suggestions??
--
James Moran <[EMAIL PROTECTED]>
Potential Technologies
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
htt
Tony,
I have downloades and installed the patches, (I think. I used patch -p0
/usr/src/zaptel/[patch], for bothe the zaptel ones, and [asterisk] for the
asterisk one).
I have addes the ukcallerid=yes to my zapata.conf, and also got
callerid=asreceived set.
The phones now ring without the screen
I just downloaded it today and the config menus just have for Firefly no
SIP or IAX2
On Thu, 2004-05-27 at 12:14, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> brian <[EMAIL PROTECTED]> wrote:
> > Just an FYI FireFly no longer works with anything but the FireFly network.
> >
> > No m
I believe it's the 'ingnorepat' option that you want. Look at the stock
extensions.conf and search for ignorepat.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Michael George
> Sent: Thursday, May 27, 2004 11:54 AM
> To: [EMAIL PROTECTED]
> S
> Your rates page still has no useful information. Can you guys at least
> put some rates up somewhere? That might encourage people to actually
> sign up. I, for one, am not going to route any calls to somebody who
> doesn't openly publish their rate schedule... seems like a recipe for a
> good
hello,
I there a problem with CVS ? My card TDM04B does not want to answer calls
on 2 ports. Strange.
B.
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Steven,
Sometimes I wonder if you have a big smile on your face when do that. Well
do you?
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 27, 2004 12:32 PM
Subject: Re: [Asterisk-Users] CVS login
> On Thu, 2004-05-27 at
Did you know that by clicking on "REPLY" instead of creating a new
message/thread many people that could have helped you won't see your
message? Those are the people that don't care about VoicePulse and so
just delete messages about Voicepulse because those messages don't apply
to them.
On Thu, 2
> Here is my problem:
> [EMAIL PROTECTED] root]# cd /usr/src
> [EMAIL PROTECTED] src]# export
> CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
> [EMAIL PROTECTED] src]# cvs login
> -bash: cvs: command not found
> [EMAIL PROTECTED] src]#
You have to install CVS on your system.
www.cvshome.org is
ignorepat => 9
- Original Message -
From: "Michael George" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 27, 2004 11:53 AM
Subject: [Asterisk-Users] generate dial tone
> The way I have my dialplan configured, an internal extension is routed
> to a different context (wi
On Thu, 27 May 2004, Tony Mountifield wrote:
> > No more SIP, No more IAX. It was a damn good IAX client... too bad its crap
> > now.
>
> Are you sure?
>
> http://www.virbiage.com/firefly/download/ still says the following:
[...]
I just download the latest version (1.7 Build 3532) and they are no
If you have usb hardware installed, you can use the ztdummy driver (part
of the zaptel bits), and you don't need usb hardware if you're using a
2.6 kernel IIRC
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of pesb
> Sent: 27 May 20
pesb wrote:
Hi there,
I need to implement a SIP Conference Server. I've saw that
asterisk has an application called meetme. But, it says that "A ZAPTEL
INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY".
Is there any other way to implement a conference server without the nee
Hi,
take a look at zaprtc (which generates the zaptel timing out of your
pc's realtime clock) or ztdummy (which uses an usb-uhci controller to
generate the timing).
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49
ztdummy will suffice. A Zaptel interface is used as a timing device for
the conference.
On Thu, 2004-05-27 at 11:58, pesb wrote:
> Hi there,
> I need to implement a SIP Conference Server. I've saw that
> asterisk has an application called meetme. But, it says that "A ZAPTEL
> INTERFA
I think if you use ztdummy that is all that is required.
Un comment in the zaptel Makefile and recompile.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of pesb
Sent: 27 May 2004 16:59
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Conference Server
H
Sorry for the downtime.
There was a configuration problem on the server that went unnoticed for a few hours.
I have no objection to mirrors, but all of the pages are dynamically generated and the
software the
wiki is currently running on doesn't provide any easy way to create and/or update
mirro
Steven,
How reliable is the current build? Do you support mySQL at this point?
Thanks,
Wojtek
- Original Message -
From: "Steven Sokol" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 27, 2004 12:24 PM
Subject: RE: [Asterisk-Users] Asterisk and PostgreSQL
> > Hi to all!!
On Thu, 27 May 2004, Chris Sullivan wrote:
> Your rates page still has no useful information. Can you guys at least
> put some rates up somewhere?
They have their rates available at
http://www.diamondcard.us/exec/voip-rep-acc-type?secRel=/secondary/corporate&priRel=/templates&secId=corporate
It
Is there any reference for the dg-104 telnet(shell)
I need to log into a remote unit and reboot it over telnet.
Its shell is not clear.
David Hickman
TSG Computer Consulting - Auctions
314-865-4752 x2
FYI ;D
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You have cvs instaled on your system?
If not then do it.
If you have it then system simply can't find cvs binary.
try
whereis cvs
Fabio Donaggio wrote:
Hi to all!!
Here is my problem:
[EMAIL PROTECTED] root]# cd /usr/src
[EMAIL PROTECTED] src]# export
CVSROOT=:pserver:[EMAIL PROTECTED]:/u
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