Hi,
Where could I download the soxmix please? I
want to mix two .gsm files into one.
Regards.
David
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Hello,
I cant compile iaxclient, because one needs to compile the new version
wiax.dll. I tried to compile it under MinGW/Cygwin, but I had the
messages like:
cc -I. -Igsm/inc -Iportaudio/pa_common -Iportaudio/pablio -Iportmixer/px_common
-Ilibspeex/include -g -O2 -DSPEEX_PREPROCESS=1 -DNEWJB
It's on the web page:
http://areski.net/areskicc-doc/
http://areski.net/areskicc-doc/AreskiCC.psql
Regards,
/Areski
On Tue, 2005-02-01 at 05:54, Gary Carr wrote:
Anyone have a copy of the DB_areskicc.psql file mentioned in the AGI tar
file for this new application?
Thanks,
Gary
For example, the first sip session will use udp 5060, but on weird
nat boxes the second sip session will get mapped to udp 5061 (or
something like that), and obviously * isn't listening on that port.
The port that shows up in sip show peers is the remote SOURCE port
and addresss.
Hi,
I'd like to trigger call recording during call. Do I have any keys that can
be pressed during call ?
I've tried this, but doesn't start anything ( I guess that a is active
only during voicemail ?):
exten = a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ;
Already recording ? if
Hi.
Do you know any IAX or SIP provider with Bangkok DID?
Sam
--
Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam
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Hi,
Does anyone have a GS101 version of the ringtone
used in the Fox show 24?
I think the ringtone is from a cisco
phone?
Cheers
Giles--
This message has been scanned for viruses and
dangerous content by
www.swiftinter.net, and is
believed to be clean.
Hello, i got configured the queues.conf and agents.conf and works well in
the first configuration for testing purposes i used
[agents]
autologoff=15
wrapuptime=5000
ackcall=no
group=1
agent = 1001,3101,Edgar de Leon
agent = 1002,,Jorge Cabrera
and when i loged in, plays a musiconhold, and
Hi,
I'd like to mark calls to get into CDR when they are received (so I'll will
be able to mark only incoming or outgoing calls without locals).
I thought that maybe doing so would work, but would kindly ask for your
opinion or maybe better advice:
a.. ${UNIQUEID}: Current call unique
Good day all
We have a few remote pbx systems running
I would like to monitor the and check that they are up and running and
working
Please Help
Altus
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Hi all,
I use isdn4linux with sip terminal.
Audio between two client sip is good but
audio between PSTN and client sip is oneway, why ?
Help me pls
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hi jay,
Jay Milk wrote:
The result can be found here:
http://www.muware.com/asterisk/
it seems as if your webserver tries to execute the .php file instead of
making them available for download...
regards
frank
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On Tue, 2005-02-01 at 12:03 +0100, Stefan Gofferje wrote:
Hi folks,
just wanted to test meetme but found out, I have none. app_meetme is
completely missing... I got the tar.gz from the link at asterisk.org.
Can somebody please point me to a non-CVS download?
You need zaptel to get
Hi,
I have downloaded files and also local versions of pwlib oh323 (both Janus
patched). Both libraries compile fine, but I get following errors on
asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention
which local libraries should be downloaded from inaccess to get everything
Hi everybody,
I'm new to this list, my name is Felipe Martins and work for a telecom
company. I'm interested in VoIP server to work as a service for my clients,
I've already configured SER to work with mysql databases and authenticate all
my users, two sip phones are already
Robert Rozman schrieb:
...
centrala:~/Asterisk/h323/asterisk-oh323-0.6.5 # make
for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory
hallo all
could anyone tell me how to get the * to send
keepalive packets over a registration "trunk"
or how to increase the amount
I'm having natting issues, (the machine is siting
behind 2 nat firewalls)
thanks
liaan
Do you Yahoo!?
Yahoo! Search presents - Jib Jab's 'Second
I use isdn4linux with sip terminal.
Audio between two client sip is good but
audio between PSTN and client sip is oneway, why ?
The usual error here is codec stuff. add the following to the sip.conf
disallow=all
allow=alaw
roy
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- Original Message -
From: Roger Schreiter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 01, 2005 1:01 PM
Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs
stableasterisk
Robert
Robert Rozman wrote:
Hi,
I have downloaded files and also local versions of pwlib oh323 (both Janus
patched). Both libraries compile fine, but I get following errors on
asterisk-oh323-0.6.5. Readme is a bit confusing since it doesn't mention
which local libraries should be downloaded from inaccess
I use GalaxyVoice and they are fine. No number portability though.
Manjit Riat wrote:
I am thinking of dumping broadvoice so I need some other VoIP providers
that have a las vegas DID and a service better than broadvoice.
One MORE
Adtran Atlas 550. I have used it in service for over 3 years and it is
ROCK SOLID!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Svensson
Sent: Saturday, January 29, 2005 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi!
Could someone give me a hand?
If I dial 200 for echo testing it works... Everytime I dial an extension ex.
505 get the error below
In this example it was from 508505 a Xlite Pro to a TA.
I believe it has something to do with the way i'm executing the command dial
but I use the standart
In your zapata.conf, where you defined your FXS port, you have to put
it in the right context so it as access to the other extensions. Just
put the same as your softphone
ex.: extension=localstations
Thank you!!That was the problem.. Context has bitten me a couple of
times, but Now I have
I just thought this link might be interesting to some of you. I know
it's m$ware but please hold back the flames.
http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp
That's fine if you want to develop an SIP phone, but if you want an
IAX one, you can take
Olle E. Johansson wrote:
Geoff Speicher wrote:
The attached patch implments a quick hack to support the Call-Info
header from the Dial() application by way of setting the CALL_INFO
variable.
This was exactly the reason I wrote the sipaddheader() function...
We can't go on adding
God save this mailling list :)
Which is the best settings for the best quality of Audio ?
I'use isdn4Linux driver and SIP client
but bad quality PSTN - Asterisk and SIP - PSTN
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[EMAIL PROTECTED] wrote:
I just thought this link might be interesting to some of you. I know
it's m$ware but please hold back the flames.
http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp
That's fine if you want to develop an SIP phone, but if you want an
IAX
Sebastian Atala wrote:
Which version of Asterisk this did work?
The patch was made against version 1.0.4, tarred, gzipped, uuencoded,
and attached to the original message.
Geoff
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I'm new to asterisk and fror a cupple of days I heave been googleing the
net for digium clones, because it's very hard for me to get a digium
card (X100P).
Does anyone Know another substitute for X100P (I know that intel based
modem with chip 537/MD3200 is working but I did not find any of
I tried to get the TDM card to work in an IBM xSeries 350 700Mhz Zeon.
What I found was that the WCTDM card does not like the 64 bit slots. If
you put the card in the 32-bit 33Mhz slot it works all day. If you move
it into a 33Mhz/64-bit slot or 66 MHz/64-bit slot, as soon as you
modprobe wctdm,
Can you give me some councils?
Sorry I can't help you, I tried myself but couldn't manage to compile it.
Somebody else on this list managed to do it, maybe he will jump in and
provide some help (Preston, are you listening ? ;)
___
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Ok, here's an odd one. I would have opened a bug on this but last time I
tried that I got flamed.. :)
Problem: When proxy requests digest challenge (SIP) Asterisk responds
normally with the exception that for some reason it changes the FROM:
(Also changes Contact: )to what's in the original TO:
-Original Message-
From: Matthew Laird [mailto:[EMAIL PROTECTED]
I excitedly installed my TDM dev kit earlier this weekend, installing
asterisk and all the kernel drivers to make it work. And it
did, it was
fantastic.
I then reboot the machine, and even after doing a modprobe
Really... I *think* I tried this already, I'll try on our old Dell 700
and see if this helps. Thanks.
-Original Message-
From: Christopher Slaght [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 01, 2005 7:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
DO you know of companies that will re-brand ip (sip/iax) phones?
thanks,
On Wed, 02 Feb 2005 00:34:23 +1100, Duane [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
I just thought this link might be interesting to some of you. I know
it's m$ware but please hold back the flames.
I'd like to open up my firwall so that I can connect my SIP phones to a
test server behind or firewall. I can configure an outside addtess to pass
traffic to the internal address of the Asterisk server. I'm not sure what
other ports need to be opened. My SIP phone will either be at my home
behind
-- voip-info.org
On Mon, 31 Jan 2005 20:21:51 -, Richard Dutton [EMAIL PROTECTED] wrote:
Hi Guys,
I know no doubt this has been covered on the list a zillion time before, but
can anyone point me to some good resources on using Asterix as a VoIP
gateway?
I would like to get two
I would be interested in this. I have Delphi 6 I believe. I'm a little
rusty, but I'd like to make a really basic client. If you have any
pointers along with that DLL, that would be fantastic.
--
Dana
On Tue, 1 Feb 2005 08:28:39 -0500, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
That's fine if
Yes, I would !!!
If you wan't, I can send you a demo of my phone.
Just drop me a private email, since this list as enough email traffic as it is
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Stefan Gofferje wrote:
Hi folks,
just wanted to test meetme but found out, I have none. app_meetme is
completely missing... I got the tar.gz from the link at asterisk.org.
Can somebody please point me to a non-CVS download?
MeetMe will not build unless you have Zaptel installed. MeetMe
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 01, 2005 1:23 PM
Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on cvs
stableasterisk
Robert
On Tue, 2005-02-01 at 14:09 +0900, Kuniyoshi Murata wrote:
Hi,
I'm thinking of setting up Asterisk for H.323 video phone clients.
Now, what is the difference between native H.323 that come with Asterisk and
Open H.323 for Asterisk ?
I can't tell you the exact differences, but oh323
I faithfully followed the instructions from:
http://www.voip-info.org/wiki-
Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
but still the message waiting indicator doesn't flash when a message is
waiting. There is a brief intermittent chirp but nothing more.
Using latest firmware 1.4.1
My
On Tue, 2005-02-01 at 13:21 +0100, Robert Rozman wrote:
By the way: use asterisk-oh-0.7.x!
But shouldn't I use 0.6.5 cause I'm on cvs STABLE ?
You are completely right. 0.6.5 for STABLE and 0.7.x for HEAD.
Regards, Bruno,
___
Asterisk-Users
what is the meaning of (cause 0).
i know that in * code it indicates an undefined cause but that's not enough.
i have many of this message in my logs.
what would be the posiible causes for this message?
i have also the same message with SIP channels...
thanks,
Paradise Dove
but still the main question mark remains:
what are the possible causes which make this warning appear
thanks!
On Tue, 1 Feb 2005 18:10:03 +0330, Paradise Dove [EMAIL PROTECTED] wrote:
but still the main question mark remains:
what are the possible causes which make this warning
On Wed, 2005-01-26 at 15:59 -0500, Paul Dugas wrote:
I've got a single IAXy installed in a little office nearby and got a call
from someone on site a finew mintues ago. Apparently they couldn't make a
call on that extension. They'd pick up the phone and get nothing; no
dial-tone.
Has
... As well it should -- LOL, it was late. I renamed them to .txt files
for the time being and will tar them later.
-Original Message-
From: Frank Sautter [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 01, 2005 5:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
There is option automon = *1 in features.conf
As I understand when *1 pressed during conversation = recording should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that feature working?
Thanks
--
Best Regards
VladK
Someone correct me if i'm wrong but I believe it's the following:
5060 UDP - SIP
8000 UDP - RTP
1-2 UDP - RTP Media (or other interval you define in
/etc/asterisk/rtp.conf)
Hope that this helped
Best regards,
Helder
- Original Message -
From: [EMAIL PROTECTED]
To:
On Tue, 1 Feb 2005, David Brodbeck wrote:
-Original Message-
From: Matthew Laird [mailto:[EMAIL PROTECTED]
I excitedly installed my TDM dev kit earlier this weekend, installing
asterisk and all the kernel drivers to make it work. And it
did, it was
fantastic.
I then
-Original Message-
Message: 4
Date: Mon, 31 Jan 2005 17:59:31 -0600
From: John Williams [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: Answering Machine Function?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL
On Tue, Feb 01, 2005 at 09:41:25AM -0500, Noah Miller wrote:
I faithfully followed the instructions from:
http://www.voip-info.org/wiki-
Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk
but still the message waiting indicator doesn't flash when a message is
waiting. There is a brief
New install, Calls are working phone to phone using gsm, ulaw or alaw
codec but when try and echo test or voicemail there is no playback.
I've tried turning on and off every codec and still no luck.
Asterisk says it's playing the sound file but I just don't hear
anything. I can't find any
-Original Message-
From: Matthew Laird [mailto:[EMAIL PROTECTED]
Hmm, found the problem, I just manually ran it again (I did
last night)
specifying the configuration file well that's annoying. I have
zaptel.conf in /etc/asterisk along with the other configs, ztcfg looks
in
- Original Message -
From: Robert Rozman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 01, 2005 3:29 PM
Subject: Re: [Asterisk-Users] Error on compiling oh323 0.6.5 on
cvsstableasterisk
-
Just tried this, same deal. A bunch of NMI errors and eventually locks
up.
-Original Message-
From: Matt Schulte
Sent: Tuesday, February 01, 2005 7:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] NMI issues...
Really... I *think* I
On Tue, 1 Feb 2005 08:08:50 -0600 (CST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I'd like to open up my firwall so that I can connect my SIP phones to a
test server behind or firewall. I can configure an outside addtess to pass
traffic to the internal address of the Asterisk server. I'm not
Hi,
I'm looking for adressbook that could easily integrate into Asterisk, so it
should:
- have agi script to search for caller id name
- have fields for notes on previous contacts (for CRM spawning of FOP)
- have web interface to edit entries easily ...
Any advice, pointers ? What is your
On Tue, 1 Feb 2005, Vladyslav wrote:
There is option automon = *1 in features.conf
As I understand when *1 pressed during conversation = recording should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that feature working?
You have to set the w or
I got mine from http://www.digitnetworks.com
On Tue, 01 Feb 2005 15:40:50 +0200, Razvan Turtureanu
[EMAIL PROTECTED] wrote:
I'm new to asterisk and fror a cupple of days I heave been googleing the
net for digium clones, because it's very hard for me to get a digium
card (X100P).
Does anyone
I am having problems getting any form of call transfer working.
I have reconfigured blind transfers to be #1 and assisted transfers to
be *2 but these are not working.
Looking at the wiki
(http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not
mention IAX so I assume I have to use the
Hi
all
I have a door entry
system that connects to an FXS Card. When the button is pressed it dials
an extension which is actually a group of all other SIP
extension.
Is there anyway you
can specify that the phones ring a different ring type when a call is received
by that FXS card
When I receive voicemail notification via e-mail I noticed that the
${VM_CALLERID) puts the IP address of the * box when callee info is
not present. Is there a way to have the field put Unkown caller in
instead of the IP address of the * box.
Kurt
___
On Tue, 2005-02-01 at 09:04 -0600, asterisk-users-
[EMAIL PROTECTED] wrote:
There is option automon = *1 in features.conf
As I understand when *1 pressed during conversation = recording
should
begin. But unfortunately it doesn't work for me.
I use CVS-HEAD-01/27/05
Does anyone has that
On Tue, 2005-02-01 at 16:26 +0100, Robert Rozman wrote:
Hi,
I'm looking for adressbook that could easily integrate into Asterisk, so it
should:
- have agi script to search for caller id name
- have fields for notes on previous contacts (for CRM spawning of FOP)
- have web interface to edit
Hi,
the new Grandstream release for the ATAs allows the setting of the FXS
impedence, the Onhook Voltage and the Polarity Reversal.
Anyone know how these should be set in Germany?
--
Best regards
Peer Oliver Schmidt
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We have an open support incident with Digium but have not yet heard back.
FWIW we have stopped selling deploying the IAXys until we have a
resolution to the problem.
Bryan what kind of power supplies do you use?
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-Matthew
- Original Message -
From: Stefan Gofferje [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 31, 2005 6:43 PM
Subject:
Quoting [EMAIL PROTECTED]:
Subject: [Asterisk-Users] mysql based adressbook with agi and web
interface ?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi,
- Original Message -
From: Gareth Blades [EMAIL PROTECTED]
...
...
I am having problems getting any form of call transfer working.
I have reconfigured blind transfers to be #1 and assisted transfers to
be *2 but these are not working.
Looking at the wiki
I have a similar door entry system and the same problem :)
I'm trying to solve it by setting CallerID to something different and
changing the ringtone to that. For the Snoms it will work, I just haven't
figured out how to do it with analog phones connected to a SPA-2000.
Cheers!
Remco
On Tue,
Hi!
With a little bit of thinking you can succeed [and there's more than one
way to do this, one alternative to the sketch below would involve MeetMe
where you'd put the called person into MeetMe and let the caller join a
little later - for recording you'd need create yet another .call file to
I THINK. When dialing 1+10 digits, I occasionally get a telco
message You must first dial a 1. When I look at the console, the
number is being sent to the ZAP channel properly. We're talking about a
couple of POTS lines on a TDM400P.
I'm thinking that it may be starting the dial too
I have a situation, where call comes in on one PSTN line, (say 2125551212), and then I want to change the Caller ID ANI to 2125551213, and then forward the call to 2125551214.
[incoming]
exten =2125551212,1,Goto(outgoing,s,1)
[outgoing]
exten = s,1,Wait(1)exten = s,2,Answerexten =
Oh! Oh! As a newbie, I can actually answer this one!
http://www.voip-info.org/wiki-Asterisk+zap+channels
Distinctive ring detection Now I hope it works, I plan to try and set
it up later this week. :)
On Tue, 1 Feb 2005, Remco Barende wrote:
I have a similar door entry system and the
have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels
Here it tells you that you can specify a wait period.
Here are some examples of complete Dial commands as they might appear in
your Dialplan
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf:
exten =
Hi All,
I'd like to develop an IAX - client.
Does somebody know where can I get the source code
for an IAX client?
Regards
César
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I found At www.ebay.com 7 $US
Max Rivera
- Original Message -
From: Razvan Turtureanu [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, February 01, 2005 11:40 AM
Subject: [Asterisk-Users] X100P Clone
I'm new to asterisk and fror a cupple of days I heave been
Hi!
standard asterisk doesnt support that. However it's in
bristuff (www.junghanns.net/asterisk) zapata.conf:
nationalprefix=0
internationalprefix=00
BTW: The same applies to chan_capi.
Cheers, Philipp
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On February 1, 2005 11:03 am, Daryl G. Jurbala wrote:
I THINK. When dialing 1+10 digits, I occasionally get a telco
message You must first dial a 1. When I look at the console, the
number is being sent to the ZAP channel properly. We're talking about a
couple of POTS lines on a
testing
Will Stowe
Systems Administrator
Promisant (USA) Inc.
email:[EMAIL PROTECTED]
Office: (770) 913-3723
Mobile: (404) 993-0526
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On Tue, 2005-02-01 at 15:58, Dan wrote:
Hi,
- Original Message -
From: Gareth Blades [EMAIL PROTECTED]
...
...
I am having problems getting any form of call transfer working.
I have reconfigured blind transfers to be #1 and assisted transfers
to
be *2 but these are not
isdn4linux is generally a good idea for initial testing and similarly a
bad idea for production.
get an hfc-pci card and use bristuff from junghanns.net if you need a
single BRI, og get some isdn hardware from the same site if you need
more BRIs. Purchase PRI stuff from digium if you need that.
Can I still use the Digium cards if I use bristuff? I bought a
Digium TE110P and will problem run into the same problem
Thanks!
Remco
On Mon, 31 Jan 2005, Klaus-Peter Junghanns wrote:
Hi,
standard asterisk doesnt support that. However it's in
bristuff (www.junghanns.net/asterisk) zapata.conf:
I have a 7960 desk phone and I'm running x-lite on my laptop. They are
both behind a NAT box so they would appear to * as being from the same
IP. I'm trying to make them ring at the same time but appear to
everyone as one extension. Is it possible to have them both register
to * as the same
I'd like to open up my firwall so that I can connect my SIP phones to
a
test server behind or firewall. I can configure an outside addtess to
pass
traffic to the internal address of the Asterisk server. I'm not sure
what
other ports need to be opened. My SIP phone will either be at my home
G'Day All,
Eyebeam has gotten my interest but I do not have a high-altitude view
of its interraction with *, therefore my questions.
I called xTEN but they preferr to talk to telcos and ISP's purchasing
hundreds of the eyebeam software... Kind-a stuck here.
I already have * happily running and
Test
Will Stowe
Systems Administrator
Promisant (USA) Inc.
email:[EMAIL PROTECTED]
Office: (770) 913-3723
Mobile: (404) 993-0526
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To
-Original Message-
snip
Does anyone know what this might be and/or an easy way to have the ZAP
channel come off-hook, delay for 1/2 second or so, and then dial?
-look at the w option to the dial command on the wiki
Exten=???,1,Dial(Zap/G?/w${EXTEN})
Jason Kawakami
Hi All,
I'd like to develop an IAX - client.
Does anybody know where can I get the source code for an IAX client?
Regards
César
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To
hi
perhaps someone should add the pix stuff to an faq
also
perhaps someone should add a DO NOT ASK THE LISTE BEFORE READING THE
FAQ somewhere
roy
On Jan 20, 2005, at 22:51, Christopher wrote:
Can anyone point me in a good direction for configuring SIP through a
PIX using 1:1 NAT. I have read
Look here :
http://www.voip-info.org/tiki-index.php?page=IAXClient
Regards,
At 18:22 01/02/2005, you wrote:
Hi All,
I'd like to develop an IAX - client.
Does somebody know where can I get the source code for an IAX client?
Regards
César
___
On Tue, 1 Feb 2005, Andrew Kohlsmith wrote:
I am seeing this too, only I'm using 100% PRI. I'm not sure how it would be
a
Zap problem since the Zap driver doesn't see any digits when you're dealing
with PRI.
Several tries later and it suddenly works It's very sporadic.
Do you
i want to know how we can add extension to mysql
database. i am using asterisk_addons and i have
checked that mysql database is connected with asterisk
here is trace
[skipping app_intercom.so]
[codec_ilbc.so] = (iLBC/PCM16 (signed linear)
I sent the same email to [EMAIL PROTECTED], and got a response
within minutes. It's now working. Previously, I sent it to
[EMAIL PROTECTED] they don't check that address,
eventhough it's listed on their site.
-ry
On Sun, 30 Jan 2005 10:15:41 -0500, Ryan Laginski [EMAIL PROTECTED] wrote:
Hi,
I
Sipuras are easy --
SetVar(ALERT_INFO=bellcore-r1)
r1 can be anything from r1 through r8, and those names, as well as
the actual ring-cadence, are configurable in the Regional tab of your
SPA-2000.
-Original Message-
From: Remco Barende [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
-Original Message-
From: Ferguson, Michael
Sent: Tuesday, February 01, 2005 11:35 AM
To: 'asterisk-users@lists.digium.com'
Subject: Messaging with * and eyeBeam
G'Day All,
Eyebeam has gotten my interest but I do not have a high-altitude view
of its interraction with *, therefore my
[EMAIL PROTECTED] wrote:
On Tue, 1 Feb 2005, David Brodbeck wrote:
-Original Message-
From: Matthew Laird [mailto:[EMAIL PROTECTED]
I excitedly installed my TDM dev kit earlier this weekend,
installing asterisk and all the kernel drivers to make it work.
And it did, it was
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