On Tue, May 20, 2008 at 12:23 AM, Lee Howard [EMAIL PROTECTED] wrote:
Andreas van dem Helge wrote:
Cisco gateway with T.38 support. That's the only real way to do faxing
through asterisk.
Although this statement has marginally more truth to it given the
SIP-only context that the original
and to the OP about the hardware, even a cheap grandsteam ATA will
work just fine... that's what I use on my personal fax machine and it
has no issues. I can't recall a time this year a fax has failed. This
is going over the public internet and then also back out to a voip
provider. We do use
I was thinking about dividing my users into different groups (contexts)
in voicemail.conf so that I could use voicemail show users for
[context] to manage them easier.
However, I found out that I should not do that because if I am using
[macro-stdexten] in extensions.conf, I will need to
On Tue, May 20, 2008 at 05:58:45PM +1000, Lee, John (Sydney) wrote:
I was thinking about dividing my users into different groups (contexts)
in voicemail.conf so that I could use voicemail show users for
[context] to manage them easier.
However, I found out that I should not do that because
Please direct me to any usefull links to help secure my asterisk server once
these ports are opened.
Thanks
Shaun
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To UNSUBSCRIBE or update
Glad to hear that, I do remember my aastra phone xfer button crashed
asterisk every time i pressed it ;)
2008/5/20 Daniel Lynes [EMAIL PROTECTED]:
Grygoriy Dobrovolskyy wrote:
Hmm, i dont like aastra really much, their transfer management is not
human
friendly ;)
Fwiw, their new 9i
Thanks, Steven.
Do both parties hear the crackling, etc?
No, just the users in the Office, and its like a background crackle that
happens with voice and silence.
This happens all the time, every call.
Called the Office number from VoIP phone to VoIP phone and the quality is
terrible.
Don't
On Tue, May 20, 2008 at 10:41:28AM +0200, Shaun Wingrin wrote:
Please direct me to any usefull links to help secure my asterisk server once
these ports are opened.
http://search.yahoo.com/search?p=secure+asterisk+server
http://www.google.com/search?q=secure+asterisk+server
Now, do some basic
OCG Technical Support [EMAIL PROTECTED] writes:
Anyone tried Asterisk with Fedora 9 (recent release)?
I upgraded my home server with Asterisk to Fedora 9 the day before
yesterday. I use the Asterisk version that comes with Fedora 9.
/Benny
___
--
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along with this voicefile.
Is it possible to record what the caller is singing?
Grt,
One way to make the system more secure would be by not opening these ports
statically in Linux iptables. I have not tested this, but Linux iptables
have shipped with ip_nat_sip and ip_conntrack_sip modules since kernel
version 2.6.18. With these modules, Linux iptables will act as a SIP-aware
NAT
Lee Howard [EMAIL PROTECTED] writes:
Note that if you have a fax machine that performs some variant of T.37
(fax-over-email) and you have an on-line service provider that is
willing to work with you... then you can rather easily get your fax
machine faxing through their service. (Which is
On Tue, May 20, 2008 at 06:46:49AM -0400, Raj Jain wrote:
One way to make the system more secure would be by not opening these ports
statically in Linux iptables. I have not tested this, but Linux iptables
have shipped with ip_nat_sip and ip_conntrack_sip modules since kernel
version 2.6.18.
On Tue, May 20, 2008 at 11:58 AM, Arjan Kroon | Mobillion
[EMAIL PROTECTED] wrote:
During this voicefile he sings along with this voicefile.
Is it possible to record what the caller is singing?
Only on non-RB or rap tracks because the lag would make the result un-funky.
On Tue, May 20, 2008 at 7:11 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, May 20, 2008 at 06:46:49AM -0400, Raj Jain wrote:
One way to make the system more secure would be by not opening these ports
statically in Linux iptables. I have not tested this, but Linux iptables
have shipped
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along with this voicefile.
Is it possible to record what the caller is singing?
Grt,
2008/5/20 Benny Amorsen [EMAIL PROTECTED] [EMAIL PROTECTED]
:
Lee Howard [EMAIL PROTECTED] writes:
Note that if you have a fax machine that performs some variant of T.37
(fax-over-email) and you have an on-line service provider that is
willing to work with you... then you can rather
On Tue, May 20, 2008 at 7:42 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along with this voicefile.
Is it
Quoting Erik de Wild: Tripple-o [EMAIL PROTECTED]:
What is the most reliable method for Asterisk
to detect the Called ID for incoming calls on
an analog line in the Netherlands?
In Holland you have to pay to receive cid info on the incoming line.
I've got that and I've tested
As a result, I just go back to put all users in [default] in
voicemail.conf.
Am I missing anything?
What do those contexts mean in your setup (beside being arbitrary
groups)?
I just want to group the mailboxes by say department rather than putting
them all under [default].
So, I could
On Mon, 2008-05-19 at 15:22 +, SVN commits to the Digium
repositories wrote:
Author: file
Date: Mon May 19 10:22:10 2008
New Revision: 117081
URL: http://svn.digium.com/view/asterisk?view=revrev=117081
Log:
Make chan_h323 work with pwlib 1.12.0
(closes issue #12682)
Iirc JerJer
Steve Totaro wrote:
On Tue, May 20, 2008 at 7:42 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along
Hi everybody,
Could anyone please answer my question.
I want to make the next scenario be possible.
1. Caller call another user.
2. Callee (called party) picks up and enters IVR menu. And then depending on
his choice he has variants to: transfer the call to another user, transfer
to voicemail,
Lee, John (Sydney) wrote:
As a result, I just go back to put all users in [default] in
voicemail.conf.
Am I missing anything?
What do those contexts mean in your setup (beside being arbitrary
groups)?
I just want to group the mailboxes by say department rather than putting
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
Could anyone please answer my question.
I want to make the next scenario be possible.
1. Caller call another user.
2. Callee (called party) picks up and enters IVR menu. And then depending on
his choice he has
Thanks a lot, that's the answer i could dream of :)
On Tue, May 20, 2008 at 3:34 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
Could anyone please answer my question.
I want to make the next scenario be
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install the
analog card asterisk will not start (ztcfg fails). I have determined it
is because of a IRQ problem and have decided to get a new server. Can
anyone suggest
Hello All,
I have an older TDM400P (~5-6 years, bought as a developer kit).
Previous versions of Trixbox worked fine with it, but I recently tried
moving to Asterisk Now, and I am not getting the card recognized.
Asterisk Now (And Trixbox) do not see the card, and insist on
configuring the
On Tue, May 20, 2008 at 07:55:32AM -0500, Cavanna, Richard wrote:
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install the
analog card asterisk will not start (ztcfg fails). I have determined it
is because of a
On Tue, May 20, 2008 at 8:55 AM, Cavanna, Richard [EMAIL PROTECTED] wrote:
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install the
analog card asterisk will not start (ztcfg fails). I have determined it
is
The first part of this is kind of off topic as it doesn't answer OP's original
question, but instead is a reply to one of the replies.
Cisco is certainly not the only option for doing T38 gatewaying with Asterisk.
I believe Asterisk 1.6 with app_fax supports T.38 origination and termination,
Do you have a spare machine you can try instead of the Dell? Even a
workstation?
Thanks,
Steve Totaro
On Tue, May 20, 2008 at 5:21 AM, Paul Goodyear [EMAIL PROTECTED] wrote:
Thanks, Steven.
Do both parties hear the crackling, etc?
No, just the users in the Office, and its like a background
On Tue, May 20, 2008 at 09:03:16AM -0400, Scott Sharkey wrote:
Hello All,
I have an older TDM400P (~5-6 years, bought as a developer kit).
Previous versions of Trixbox worked fine with it, but I recently tried
moving to Asterisk Now, and I am not getting the card recognized.
Asterisk Now
Matt Watson wrote:
I believe Asterisk 1.6 with app_fax supports T.38 origination and
termination, that is not gatewaying, however if origination and termination
are already there, gatewaying should be fairly trivial to implement. I
haven't actually tested 1.6 using T.38, however I have
On Tuesday 20 May 2008 07:10:59 Lee, John (Sydney) wrote:
As a result, I just go back to put all users in [default] in
voicemail.conf.
Am I missing anything?
What do those contexts mean in your setup (beside being arbitrary
groups)?
I just want to group the mailboxes by say
T.38 gateway is a totally different problem than T.38
origination/termination. They share very little code, and almost none of
their design.
Regards,
Steve
Well,
It turns out their SIP provider doesn't support the T.38 protocol for faxing.
Their statement is if you really need it, use ulaw and
I would suggest actually looking at the arguments for the queue command.
There is an option to play a ring instead of hold music.
To look at the syntax for any command in asterisk, from the cli type
show application command name.
So for you:
show application queue
--
Thank you and have any
We are running Asterisk 1.4. Does anyone have thought about how to pass
RDNIS from a callmanager to a just an Asterisk 1.4 voicemail server?
I have tried to use CALLERID(rdnis) with a NoOp statement prior, but I
get confused by the syntax of whether or not to use the SET in front of
the
Perhaps your carrier has changed the way they handle outbound caller id.
Could they be blocking it somehow all of a sudden?
Might be worth a phone call?
steve
On Thu, 2008-05-08 at 08:29 +0100, Tim Guy wrote:
Thanks for the heads up again guys.
Still no go.
It's a ISDN30 PRI on
Joseph L. Casale wrote:
It turns out their SIP provider doesn't support the T.38 protocol for faxing.
Their statement is if you really need it, use ulaw and AstraFax? I don't
understand
how AstraFax makes a difference in the process?
It doesn't make a difference. uLaw over SIP/UDP still
Hi,
I am having trouble with Polycom forwards and Asterisk. Basically, I have
no clue on how to force callerid or even custom variables (set using SetVar
in the sip.conf file) on the transfered call.
For example, I set a variable called var_a to foo. When the call comes
in, the variable
On Tue, May 20, 2008 at 12:49 AM, Kevin Smith [EMAIL PROTECTED] wrote:
I almost hate to admit this...but I'm still running Asterisk 1.2 on
Fedora 4 :D
Same here (FC4 + Asterisk 1.2.10 + TDM + TE + iaxmodem + Hylafax).
Working flawless for ages. Old * because I had problems with upgrading
zaptel
Andreas van dem Helge wrote:
So why don't you just disable reinvite?
Using 1.4.15 here with no issues with MixMonitor. Then again I've
*ALWAYS* disabled reinvite because it never works for me.
Yeah, I've got some older servers running that do not have this same
problem. I believe it's
Richard Cahilig wrote:
Hi,
I am also experiencing problem with MixMonitor in Asterisk 1.4. Most of
our recording is overlapping. How can I fix this problem?
Do you mean overlapping like one side is delayed so you hear the two
parties talking on top of each other?
That's another thing I've
On Tue, May 20, 2008 at 07:55:32AM -0500, Cavanna, Richard wrote:
I am having a issues with adding a analog card to my dell 2800. I
already have a t1 card installed and running fine but when I install
the
analog card asterisk will not start (ztcfg fails). I have determined
it
is because of a
On Tue, 2008-05-20 at 10:55 -0400, Mike wrote:
Hi,
I am having trouble with Polycom forwards and Asterisk. Basically, I
have no clue on how to force callerid or even custom variables (set
using SetVar in the sip.conf file) on the transfered call.
For example, I set a variable
Benny Amorsen wrote:
Lee Howard [EMAIL PROTECTED] writes:
Note that if you have a fax machine that performs some variant of T.37
(fax-over-email) and you have an on-line service provider that is
willing to work with you... then you can rather easily get your fax
machine faxing through
The upload of the java applet is a little slow but very very slick :-) -
nice use of asterisk for 'non voice specific purposes'.
I'm in this room for the next 10 minutes if people want to stress test
it out to see how well it scales :-)
http://www.twiddla.com/19035
Regards,
Dean
Lee Howard wrote:
As I tried to indicate in my first reply... I would encourage you to
order an analog phone line for that fax machine. There are other
options, but in most cases the customer is happy to pay the line charge.
When you go the analog POTS line route it just works. No messing
Hey everybody,
I'm still having issues with this system. The phones won't stay
registered for more then a few minutes. They're bouncing up and down.
I'm able to ping the phones just fine. What I've done so far:
Power cycled all phones and verified
Power cycled all switches
Checked the ARP
Hi All,
We have a Cisco CME linked to our Asterisk PBX (named 'epstein'). Off
said PBX we have numerous other PBX's, some IAX and some SIP. On a
call placed from CME (SIP) to 'epstein' it all works fine except for a
few quirks.
When calling through epstein to an IAX peer we get '100
Thank you all for your input. Currently, nothing has improved the
dropped call rate by more than .2%, leaving me at 1.8% dropped calls
still..Luckily, our switch back to PRI is due anytime in the next day or
so..
Sherwood McGowan
___
-- Bandwidth
On Tue, 2008-05-20 at 11:51 -0400, Doug Lytle wrote:
Hey everybody,
I'm still having issues with this system. The phones won't stay
registered for more then a few minutes. They're bouncing up and down.
I'm able to ping the phones just fine. What I've done so far:
Power cycled all
I am having trouble with Polycom forwards and Asterisk. Basically, I
have no clue on how to force callerid or even custom variables (set
using SetVar in the sip.conf file) on the transfered call.
For example, I set a variable called var_a to foo. When the call
comes in, the variable
Did you try _var_a? Iirc you need to prepend it with an underscore to
make the variable persistent.
Forget my previous email, it didn't quite work that simply but I tweaked my
dialplan and you had the right solution.
Thank you,
Mike
___
--
On Tuesday 20 May 2008 11:26:34 Mike wrote:
I am having trouble with Polycom forwards and Asterisk. Basically, I
have no clue on how to force callerid or even custom variables (set
using SetVar in the sip.conf file) on the transfered call.
For example, I set a variable called var_a
Mike wrote:
I am having trouble with Polycom forwards and Asterisk. Basically, I
have no clue on how to force callerid or even custom variables (set
using SetVar in the sip.conf file) on the transfered call.
For example, I set a variable called var_a to foo. When the call
comes in, the
Patrick wrote:
Not sure if this helps but iirc I've seen checksum issues on an Asterisk
DHCP box that I was able to get rid of by turning off some of the
Thanks Patrick.
I've narrowed the issue down a bit, I put 2 Polycom phones and the phone
system on it's own switch, the phones
Remove the qualify= option from sip.conf. Also make sure the DISABLE
CDP in the Polycom's boot menu.
Doug Lytle wrote:
Hey everybody,
I'm still having issues with this system. The phones won't stay
registered for more then a few minutes. They're bouncing up and down.
I'm able to ping
Tzafrir Cohen wrote:
On Tue, May 20, 2008 at 09:03:16AM -0400, Scott Sharkey wrote:
Hello All,
I have an older TDM400P (~5-6 years, bought as a developer kit).
Previous versions of Trixbox worked fine with it, but I recently tried
moving to Asterisk Now, and I am not getting the card
I'd like to take a few moments to introduce myself and the new role
in which I'll be working for Digium to further the Asterisk project
and environment. As you may know, Digium plays a key part in
assisting with the development of the Asterisk project, and so I am
pleased to be working for
Interesting... I really don;t know the T.38 protocol other than what it does.
How it goes about doing it I haven;t really gotten into.
I would of thought that gatewaying would of (essentially) be a bridge between a
termination and origination action. However that is just completely sort of
You might want to see if you can change the IRQ assignments in your servers
bios (might have to turn off the PNP OS Installed option if you have one)
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL
PROTECTED]
Sent:
Yes, thats the problem I encountered so far with Asterisk version 1.4
On Tue, May 20, 2008 at 11:03 PM, Trevor Peirce [EMAIL PROTECTED]
wrote:
Richard Cahilig wrote:
Hi,
I am also experiencing problem with MixMonitor in Asterisk 1.4. Most of
our recording is overlapping. How can I fix
On Tue, May 20, 2008 at 7:41 PM, John Todd [EMAIL PROTECTED] wrote:
I'd like to take a few moments to introduce myself and the new role
Hi John,
Like Jared, you need no introduction to most of us, you are a pillar
of the asterisk community. When I first heard of asterisk, the first
information
Eric Wieling wrote:
Remove the qualify= option from sip.conf. Also make sure the DISABLE
CDP in the Polycom's boot menu.
That didn't help and CDP is off by default, the phones still couldn't
receive/send calls when in this state. I've sent an employee out to
grab a replacement NIC.
On Tue, May 20, 2008 at 7:41 PM, John Todd [EMAIL PROTECTED] wrote:
I'd like to take a few moments to introduce myself and the new role
Hi John,
Like Jared, you need no introduction to most of us, you are a pillar
of the asterisk community. When I first heard of asterisk, the first
John,
Welcome to the new position!
From all the discussions we've had in the advisory council and before
that, I know that you have been in that position for a long time, the
difference is that Digium started paying you now...
Looking forward to the future of the Asterisk project :-)
Doug Lytle wrote:
Eric Wieling wrote:
Remove the qualify= option from sip.conf. Also make sure the DISABLE
CDP in the Polycom's boot menu.
That didn't help and CDP is off by default, the phones still couldn't
receive/send calls when in this state. I've sent an employee out to
On Tue, May 20, 2008 at 11:12:17AM -0700, John Todd wrote:
Thanks! I've got my work cut out for me. :-)
Very likely. Allow me to give you the welcome traditional to technical
newsgroups and mailing lists:
Welcome To The Jungle.
Cheers,
-- jra
--
Jay R. Ashworth Baylink
I'm trying to get asterisk to proxy h263 for a video call, but not having
any luck. I have posted a full call trace here:
http://pastebin.com/d330aecb5
While watching a full packet dump on the asterisk node, I can see the h263
coming in from the clients, but it never leaves (asterisk never
Eric Wieling wrote:
Doug Lytle wrote:
Based on the SIP poke message you pasted in an earlier message, the
qualify= option you used is virtually guaranteed to cause SIP poke problems.
Understood, wouldn't it also indicate that, when putting 2 phones and
the phone system on it's own little
Perhaps seeing some of your dial plan (such as the macro, etc) would
help not only me, but also others, because maybe I am just not following
you.
Off the top of my head there are a few things you could do..but again,
it depends on how your dialplan is set up and how you access the macro.
One
congratz!
Zoa
John Todd wrote:
On Tue, May 20, 2008 at 7:41 PM, John Todd [EMAIL PROTECTED] wrote:
I'd like to take a few moments to introduce myself and the new role
Hi John,
Like Jared, you need no introduction to most of us, you are a pillar
of the asterisk community.
2008/5/17 bilal ghayyad [EMAIL PROTECTED]:
So no way to discover the status of FXO if a cable
pluged or not?
Did you read my previou msg
Hookstate (FXS only): Offhook --Cable plugged
Hookstate (FXS only): Onhook --Cable unplugged
--
PicoStreamer - the real WEB live
Welcome!
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I have customer in the midwest that can't navigate my IVR Menus.
I have two PRIs coming in to an Asterisk box. I enabled DTMF logging to the
console and am able to see my own tones when I call in, however when this
user calls in I see nothing. Is there a deeper level of debug I can do?
Thanks,
VMukti Open Source Edition edition installed (sort of..) on a
clean Win2k3 server box (what a pain!!) will keep this thread updated
or start a new one.
On Tue, May 20, 2008 at 11:32 AM, Dean Collins [EMAIL PROTECTED] wrote:
The upload of the java applet is a little slow but very very slick
We are having the same issue here as well and zaptel / zapata is configured properly because we used the same card for 1.4.X . All worked properly I am going to watch this issue closely on the board for solutions.VA
Original Message
Subject: Re: [asterisk-users] AsteriskNow: No
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm not
getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel 1.4.10
___
-- Bandwidth
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that channel, I need to busy the line so calls
will
You could simply short-circuit the two wires of the line. The telco will
interpret that as a busy line.
Other than that, you could do this on extensions.conf:
[context]
exten = s,1,Answer()
exten = s,n,Busy()
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
-
SIP poke does NOT just measure network latency. It also measures the
PHONE latency. Asterisk sends a SIP OPTIONS packet to the phone, the
phone responds, Asterisk measures how long it took. Most phones seem to
make responding to OPTIONS packets a low priority. A phone busy doing a
The problem is that I do not have physical access to the server. The
other problem with that solution is that the first person to dial the
main number will always get a busy tone and only then can someone else
get to another line.
I need to leave the line offhook until further
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access to the server. The
other problem with that solution is that the first person to dial the
main number will always get a busy tone and only then can someone else
get to another line.
Thank you. Unfortunately the phone Company in Mexico is not very
helpful when it comes to those services.
On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote:
On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote:
The problem is that I do not have physical access to the
Just create an extension like this:
[busyoutline]
exten = 111,1,Answer()
exten = 111,n,Busy()
then drop a .call file like this:
Channel: Zap/1/111
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: busyoutline
Extension: 111
Priority: 1
The above should work, however keep in mind if the
On Tue, 20 May 2008, Carlos Chavez wrote:
Is there a way to busy out a Zap channel? I have a customer who is
having problems with a line connected to a TDM800 card and we would like
to busy out that line. Since that line is the head of the hunt group I
cannot simply disable that
The Asterisk.org development team has released Asterisk version 1.4.20.
This release contains a large number of bug fixes over the previous release.
For a full list of changes, see the ChangeLog included in the release.
http://svn.digium.com/view/asterisk/tags/1.4.20/ChangeLog?view=markup
Joe Pukepail wrote:
Is there a way to see error counts on the T1 of a PRI? Hooked up to
asterisk via a digium TE122. Looking for something to make sure I'm
not getting any CRC, framing or other errors on the T1.
Using asterisk 1.4.19 and zaptel 1.4.10
Go on ebay and buy an ADC Kentrox
On Tue, May 20, 2008 at 2:14 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, May 20, 2008 at 8:55 AM, Cavanna, Richard [EMAIL PROTECTED] wrote:
...
Cards I have installed:
Digium TE205P - 5v
TDM410
I hear rave reviews about Supermicro but no personal experience.
I like
I haven't been following the conversation, but why don't you use
searchcontexts=yes in voicemail.conf? As long as you don't specify a
particular context when calling Voicemail, it will look through all
contexts
until it finds a matching mailbox.
Tilghman, you are spot on!
As it turns out,
Hi,
Why not use MixMonitor(), so you have a single file with the singer
and the music?
Thanks.
Andy
On 5/20/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a
Hello All
We are going to manage answering service of different clients in Pakistan.
We need US
toll free number. Please provide us rates.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL
Yes, Thanks, Monitor() was the solution.
It works perfect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Kuo
Sent: woensdag 21 mei 2008 5:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] karaoke
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