I have a list of all my local (free) on my POTS prefixes. Is there a way to integrate that so * decides
if it is going to use iax or POTS? There is about 60 prefixes.. 1831-XXX
extension.conf clipping help
--
Gafachi.com - referal code hunter81
instant iax termination - 2 cents a minute
Also
Hi,
i am using iax client and when i try one of my extension that play MusicOnHold()
it give me this error, who have an idea about this
- Executing MusicOnHold([EMAIL PROTECTED]/1, ) in new stack Jun 4 15:36:37
WARNING[1217602880]: chan_iax2.c:2838 iax2_send: timestamp is 0?
Jun 4 15:36:37
USE SUBJECTS!!!
On Wed, Jun 02, 2004 at 06:18:20PM -0700, Deepak Malhotra wrote:
Hello
I have an interesting situaltion and not sure if I am doing something wrong or
it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on
Rhino's Zap Channels. If i pickup analog
I will keep in mind, but still no solution.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 03, 2004 1:13 AM
Subject: Re: [Asterisk-Users] (no subject)
USE SUBJECTS!!!
On Wed, Jun 02, 2004 at 06:18:20PM -0700, Deepak Malhotra wrote:
Hello
Hello
I have an interesting situaltion and not sure if I am doing something wrong or
it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on
Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any
number , I am getting extra ring after hangup and if i
Wondering if anyone tried to port Asterisk to the Linksys 54G
OpenSource platform?
I am planning to try to port some of the Asterisk code to
that platform and if any once already tried I would like to get in touch with
them . I am thinking on porting the protocol and some other
was wondering if anyone could give us a run through
an explanation of the wiki and other examples of connecting to iptel's sip
express router using asterisk pbx so i can understand better the call
processing ..
given the example i work from on john todd's
www.loligo.com site ;
exten =
USE SUBJECTS!
On Tue, May 25, 2004 at 03:51:17PM +0100, Graham Turner wrote:
was wondering if anyone could give us a run through an explanation of the wiki and
other examples of connecting to iptel's sip express router using asterisk pbx so i
can understand better the call processing ..
Hello
I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card
and one USB one port FXS card. I can modprobe wcusb but ztcfg always return
ZT_CHANCONFIG failed on channel 2: No such device or address (6)
error message.
Also unable to config outgoing call using SIP
Any working examples of configuration files is highly appreciated.
http://www.voip-info.org/wiki-Asterisk
:)
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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You'll also have to modprobe the x100p
/sbin/modprobe -k wcfxo
/sbin/modprobe -k zaptel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, May 16, 2004 9:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] (no subject)
Hello
i did that bit no luck.
- Original Message -
From: Todd Lieberman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 16, 2004 6:47 PM
Subject: RE: [Asterisk-Users] (no subject)
You'll also have to modprobe the x100p
/sbin/modprobe -k wcfxo
/sbin/modprobe -k zaptel
What does 'cat /proc/interrupts' tell you?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Deepak
Malhotra
Sent: Sunday, May 16, 2004 11:09 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] (no subject)
i did that bit no luck.
- Original Message
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] recommend a Linux based TFTP server
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box?
Thanks in advance
Robb
Do you Yahoo!?Yahoo! Movies - Buy advance tickets for 'Shrek 2'
Hi Roger!
the databse i have created has all the immaginable permission.if i try to
access it with dbtools with the
username and the password of asterisk from another pc inside the network, i can
see the table
and i can read/write. Only * can't write into the table. So,cecking the conf
file
Il 21:38, venerdì 30 aprile 2004, Philipp von Klitzing ha scritto:
Hi!
I try to connect an MGCP device(Terayon) to asterisk. I have found many
example BUT the Terayon always return error 510 ! Verb:'510'
Identifiers :'2' Endpoint: 'Error' Version'(null)'
1. Which version of Asterisk
Does anyone now an Asterisk consultant in Atlanta?
Bobby
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On Mon, 2004-04-19 at 14:22, Bobby Whitley wrote:
Does anyone now an Asterisk consultant in Atlanta?
Start Here,
http://www.voip-info.org/wiki-Asterisk+consultants+USA
--
Steven Critchfield [EMAIL PROTECTED]
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Does anyone now an Asterisk consultant in Atlanta?
1. Use the subject line - it's there for a reason. (no subject) won't draw
too many people to read your message.
2. The wiki is your friend. See the URL below. There's no one listed for
Atlanta, maybe that's why you're asking.. but see point 1
I am getting ready to do my first build on this product. It's just for use as an overglorified answering machine right now and I will most likely play with some of the SIP functionality.
My big question though,is how much disk spacedo messages take up on the system? Are there any published metrics
Hi!
I have a little big problem here. I have an gateway(asterisk,working as a H.323
- SIP gateway) conected to a gatekeeper (two different servers), and also a
gateway (cisco - PSTN) conected to the same gatekeeper. When I make a call from
the gateway(cisco) to a sip phone, the phone rings, but
Hi,
Mireia Munoz de jesus wrote:
Hi!
I have a little big problem here. I have an gateway(asterisk,working as a H.323
- SIP gateway) conected to a gatekeeper (two different servers), and also a
gateway (cisco - PSTN) conected to the same gatekeeper. When I make a call from
the gateway(cisco) to a
unsubscribe
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When my snom200 receives an inbound SIP external sip call,
it somehow rejects the call and with a busy tone. The debug shows the
following error:
channel.c:1142 ast_read: Exception flag set on
'SIP/sipphone-7796', but no exception handler
what does this mean and how can I debug it
Has
anyone had any luck using a 7910 with SIP image.
Some
information I found says 7910 is skinny only, other info suggests the 7910 may
take the 7960 sip image.
Can
anyone offer their experience ?
Cheers
Peter
Search the archives.
On Tue, 2004-03-30 at 19:00, Peter Mitchell wrote:
Has anyone had any luck using a 7910 with SIP image.
Some information I found says 7910 is skinny only, other info suggests
the 7910 may take the 7960 sip image.
Can anyone offer their experience ?
Dialing in from the pstn to sip phones (x-lite softphone on winders and
a grandstream handytone-286 ata), I hear the sip phone ring a few times,
I ran into the same thing with Cisco 7960. Looks like the logic in the
sip channel has changed recently.
Add a ,r to the end of your Dial statements in
Hi guys,
Has anyone played around/got it to work app_prepaid.c?
(http://www.voip-info.org/wiki-Asterisk+callingcard)
With what data do you populate the database with cards, providers,
tariffs, tariffrates etc.. (format) to make it work. What is the
meaning/purpose of each table/field?
I am
: Alexander Romanov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 8:35 PM
Subject: [Asterisk-Users] (no subject)
Hi guys,
Has anyone played around/got it to work app_prepaid.c?
(http://www.voip-info.org/wiki-Asterisk+callingcard)
With what data do you populate the database
Hi all.
If I want to use the * only as a GW to PSTN and allow only one external
proxy to place calls. how is the smartest way to do this ?
I dont want the world to be able to do invites only a specific IP,
in this case my proxy that handles all the users.
/Mike
traces.
Thanks all
Rohde
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 10:23 PM
Subject: Re: [Asterisk-Users] voicemail issue
How do you start asterisk? using safe_asterisk? or what cli options do you give it?
bkw
I am a new asterisk user and I love what I see so far.
I have a question about distinctive ring though.
In my situation, we have 1 phone number for voice calls and one for faxes.
They share the same line, and right now I use vgetty with mgetty+sendfax and
VOCP to deal with calls and faxes.
Vgetty
T. Chan wrote:
I recently came across DynEXTENdb, a way to be able to include
thousands of Extensions (routes). In my application which is VOIP, we
need to include more than 50,000 area codes due to the USA LATA
routing, and there is simply no way to do that with extensions.conf.
The way
We are new in Asterisk - I was wondering if someone can recommend a good
phone sets to use with Asterisk in office environment. We need about 20
sets.
Also - What can we use for the receptionist phone?
Thanks,
Aram Ter-Martirosyan
Hello,
My clients usually work the regular 8 - 5 day, however they
would like to have control of the night time context.
Is there any way, say a receptionist, can dial a 4 digit extension, to
toggle on/off the night time context?
Thanks in advance,
Brent
On Wed, 2004-01-07 at 10:38, Brent Franks wrote:
Hello,
My clients usually work the regular 8 - 5 day, however they
would like to have control of the night time context.
Is there any way, say a receptionist, can dial a 4 digit extension, to
toggle on/off the night time context?
Hi!
How can I make * ring one phone then if no answer
Go to a different extension ??
Read the handbook draft which is to be found on www.asterisk.org.
Or read the Wiki and search for the description of the application DIAL.
*sigh*
Cheers, Philipp
Dear All,
I have had a
problem that I have posted before, the asterisk kept crashing on me. I have
thought that may be before the problem is resolved, I could try to implement a
cronjob to run /usr/sbin/safe_asterisk, and if Asterisk is not running at that
time, it will start it
How can I make * ring one phone then if no answer
Go to a different extension ??
[in_andrew]
exten = s,1,Dial(IAX2/[EMAIL PROTECTED]/s,20,Ttr)
exten = s,2,SetAbsoluteTimeout(900)
exten = s,3,Dial(${FXOTRUNK}/5551212,20,Ttr)
exten = s,4,Hangup
exten = h,1,Hangup
exten = t,1,Hangup
This is the
Hi all
How can I make * ring one phone then if no answer
Go to a different extension ??
Any help always appreciated
Regards Mick
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exten = s,1,Dial(Zap/2,10) - (ring Zap2 for ten seconds)
exten = s,2,Dial(Zap/3,10) - (if no luck, ring Zap3
for ten seconds)
exten = s,3,VoiceMail(u)- (if no luck, leave a message
on voicemail box )
Kind regards,
Matt Riddell
Surecall New
]
Subject: Re: [Asterisk-Users] (no subject)
exten = s,1,Dial(Zap/2,10) - (ring Zap2 for ten
seconds)
exten = s,2,Dial(Zap/3,10) - (if no luck, ring Zap3
for ten seconds)
exten = s,3,VoiceMail(u)- (if no luck, leave a message
on voicemail box
Hence why I ask for a company name. Small correction to your post, if it's
distributed to anyone, the source must be available to EVERYONE.
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 5:48 PM
Subject: Re: [Asterisk-Users
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 3:48 AM
Subject: Re: [Asterisk-Users] (no subject)
Well if it links to asterisk and or used any of its code as a base it
can't be sold without a comercial lic. for asterisk. Thats my
understanding of the GPL. If its sold
Nicolas Gudino wrote:
I'm not a GPL expert, so I have a few questions: Does an AGI script needs to
be distributed in source form? Maybe this application/script is using
Asterisk unmodified. They can sell just their AGI scripts and provide only
asterisk with full source?
An AGI script does not
Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 5:48 PM
Subject: Re: [Asterisk-Users] (no subject)
Well if it links to asterisk and or used any of its code as a base it
can't be sold without a comercial lic. for asterisk. Thats my
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 10, 2003 10:28 AM
Subject: Re: [Asterisk-Users] (no subject)
Adam Hart wrote:
Hence why I ask for a company name. Small correction to your post, if
it's
distributed to anyone, the source must be available to EVERYONE.
IANAL
Hi,
Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that covers
needs in that area: scratch number generation, call
termination via least-cost route (i.e. multiple
termination
Are they apps that load into asterisk? or agi scripts?
bkw
On Tue, 9 Dec 2003, [iso-8859-1] Kita B. Ndara wrote:
Hi,
Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that
Is there a company website? or just a free yahoo email address?
- Original Message -
From: Kita B. Ndara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 4:01 PM
Subject: [Asterisk-Users] (no subject)
Hi,
Our firm has developed two applications that I
? or just a free yahoo email address?
- Original Message -
From: Kita B. Ndara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 09, 2003 4:01 PM
Subject: [Asterisk-Users] (no subject)
Hi,
Our firm has developed two applications that I
thought might be of interest
Hi,
First at alll, I beg your pardon because maybe I explained bad my questions
(because my low level english)
I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2
compiled and installed.
I have modules alsa 0.9.8 compiled and installed
My PC has an audio card ac97
On Mon, 17 Nov 2003 20:30:09 -0500 (EST), Bob Bevins [EMAIL PROTECTED]
wrote:
-- Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011 format: wav49,
0x80de8b8
-- x=1, open writing:
/var/spool/asterisk/voicemail/local/31/INBOX/msg0011
Hi guys,
I am having a problem that I can't find an answer on digium and or the list.
When a call covers to vm, and starts to record the message I get the
following on the console. I am running redhat 9, last nites source, with
one X100P and a TDM400 with three extensions. I have tried to find an
7083533
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how do I prevent people from calling as soon as I restart the * server ? cos' this
will result (I assume)
in pri channels getting blocked. Because of those few calls that's taken during
restart results in
those few pri channels not to get properly restarted. I need something like 1~2
minutes
Brian Hello, I resolved my echo issue using grandstream/estara
etc etc Brian sip phones and wcfxo interfaces from digium. I
swapped out my Brian via kt400 based msi kt4vl motherboard for an
asus p4pe? i845? Brian based motherboard and the echo has
completly gone away along Brian with
options to try.
- Original Message -
From: Dana Dominiak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, October 17, 2003 6:10 PM
Subject: [Asterisk-Users] (no subject)
Brian Hello, I resolved my echo issue using grandstream/estara
etc etc Brian sip phones and wcfxo
Hi,
Checked out latest CVS and no sound from Playback, Background, MOH or
bridged channels. mpg123 is active but no sound.
Master
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Hello,
Can Asterisk perform as a H323 Gatekeeper?
Here is my scenario:
I have a customer that has a calling card program that will be
transmitted as VOIP from a Cisco 5300 in Hong Kong and terminated here
in North America. The catch is that, the termination is being handled
by a third party
Dear All,
I am going to deploy a VOIP
network here in Canada with nodes all over town. This is for long distance
services and hence would need a good reliable solution.
I have looked into * and am
very interested in it with all the value-added features as well as its
capability to do H323
]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 1:26 PM
Subject: RE: [Asterisk-Users] LineJack + Asterisk HELP!
-Original Message-
From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 11:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
From what I see this *IS* a problem with the CVS code...
as a quick fix I suggest using the zaptel code from august 18th 2003 since that is
known to work (I'm using it after having the same problems as you)
It's kinda strange if this isn;t regarded as a bug, as Digium have then EOL'd some of
Yep, it probably will not work with your motherboard. You might try
setting -DNO_CALIBRATION in the Makefile, then running 'make clean
all install' and trying again (this has worked for some people).
Failing that, try it with a different motherboard.
-Tilghman
This is a CODE issue not a
On Saturday 13 September 2003 11:09, Andy Powell wrote:
From what I see this *IS* a problem with the CVS code...
as a quick fix I suggest using the zaptel code from august 18th 2003
since that is known to work (I'm using it after having the same
problems as you)
It's kinda strange if this
It's kinda strange if this isn;t regarded as a bug, as Digium have
then EOL'd some of their cards and not told anyone, while continuing
to sell them...
Compare revision E to revision C of the card. Revision C is no longer
being sold by Digium.
This may be true, however, they were being sold
Andy Powell wrote:
This may be true, however, they were being sold in May of this year and I don;t
expect a piece of hardware to have a lifespan of 3.5 months!
From what I hear revision C cards are green and revision E cards are blue. It certainly
also sounds like some people were getting the
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
You may find more
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including
On Friday 12 September 2003 02:37 pm, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters,
Is it this maybe?
Communication controller: Tiger Jet Network Inc. Model
300 128k
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What does 'dmesg' says ?
Martin
On Fri, 12 Sep 2003, James Sharp wrote:
On Fri, 12 Sep 2003, Jim Paraschou wrote:
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
hello all
well while trying to make a call from gnophjone registered with IAXTEL to
another phone registered with iaxtel, we get disconnected everytime a call
is made, and the following message log is generated
17006383019 is a phone number
Using: [EMAIL PROTECTED]:5036/[EMAIL PROTECTED]
Trying
What if someone adds your number to that list ?
Someone would have to moderate it.
regards
Martin
On Tue, 5 Aug 2003, McAughan, Matt wrote:
Does anyone keep a known telemarketer caller id database? If not has anyone
proposed an Asterisk community project to share this information? Sort
: Tuesday, August 05, 2003 4:13 PM
To: '[EMAIL PROTECTED]'
Subject: Re: [Asterisk-Users] (no subject)
What if someone adds your number to that list ?
Someone would have to moderate it.
regards
Martin
On Tue, 5 Aug 2003, McAughan, Matt wrote:
Does anyone keep a known telemarketer caller id
Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls...
On Tue, 5 Aug 2003, McAughan, Matt wrote:
Does anyone keep a known telemarketer caller id database?
Here it is:
CALLER UNKNOWN
PRIVATE
:)
Most CLID's come up Unknown.
--
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22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve Sobol, Proprietor
ÐÁÎÁÓÏÎÉË Ó ÁÓÔÅÒÉÓËÏÍ.
Date: Wed, 30 Jul 2003 20:06:17 +0400
From: Pavel Zheltouhov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232
Reply-To: [EMAIL PROTECTED]
I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk
Hi,
I am a new to Asterisk. I am looking for a cheap solution for PC-to-Phone
call serving one or two cities. Can anyone provide me with pointers to
architecture documents/other documents from where I can start. I am not
new to VoIP.
Regards,
Deepak Mittal
Newbie question, please excuse me for this
one. If an admin adds and extension in the voicemail.conf file will
asterisks read from .conf files dynamically? Or does the asterisks daemon
need to be restarted? I guess this question pertains to all .conf
files. Also is there support for MySQL?
I'm looking at getting the Dev light applications from digium
and I have 2 Createive Labs voip blasters. The voip blaster supports the
G.723.1 codec. After looking at Gnome meeting it does not talk unless you have a
quicknet card for it. Can I make calls using asterisk and the digium cards to the
Is this me or what?
-- Playing 'demo-congrats'
-- Executing MeetMe(H323:996, ) in new stack
-- Playing 'conf-getconfno'
== Parsing '/etc/asterisk/meetme.conf': Found
WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open
pseudo channel
-- Playing 'conf-invalid'
I don't know what that is, so probably not. Is that a conference type
board? Is there a way to make conferencing work or to assign an
extension to a h323 connection?
Thanks
On Mon, 2003-06-23 at 23:42, Jeremy McNamara wrote:
Do you have a Zaptel device in this machine?
Jeremy McNamara
You must have a Zaptel device installed in your computer or load
ztdummy module to get conferencing to work...
Jordan Peterson wrote:
I don't know what that is, so probably not. Is that a conference type
board? Is there a way to make conferencing work or to assign an
extension to a h323
Hey all,
I have a E1 setup with a E400P digium card. Everything works just great
except for the callerid. When i make an outgoing call via the E1 to a
hardphone somewhere, all i get is private number. In my zapata.conf
however , i have defined the following:
context=localE1
group = 1
Try to explicitly add this line
,1,SetCallerid,(somename 12345)
,2,Dial,Zap/g1/${phonenumber}
regards
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
Hey all,
I have a E1 setup with a E400P digium card. Everything works just great
except for the callerid. When i make an outgoing call
Hi everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that
On Tue, 2003-06-10 at 16:22, Johnny Witt wrote:
Hi Asterisk-Users
Ive been reading about the Asterisk project (all that I could get my
hands on J ). It sound to good to be true. But Ive got some questions
which I havent found a answer to anywhere :
1) Can I use Asterisk as a Call
hi!
I wanna do some arithmatic operations (addition and substraction -integer
operation) inside extensions.conf. Is there a simple way to do this. If I do
yy = ${xx} + 1 // say xx is initialized to '0'
the resulting yy will show
0 + 1
Obiviously not the result I need. Any help
denzel:
read the asterisk/README.variables
exten = s,1,SetVar(v2=0)
exten = s,2,Playback(beep)
exten = s,3,SetVar(v2=$[${v2} + 1])
exten = s,4,GotoIf($[${v2} 2]?t|1:*|1)
line 3, increments the value of the variable, we use it to loop a context
for a limited number of times, etc...
- wasim
For those of you running OSX, a new h323 client was released. Haven't
set up h323 yet, so I can't vouch for it.
http://xmeeting.sourceforge.net/
--Mike
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it is possible to use musiconhold. i added
exten = s,5,SetMusicOnHold,default
exten = s,7,dial,SIP/michaelSIP/frank|15|m
to the extension conf.
the logfile looks good, i think
-- Called michael
-- Called joern
-- Started music on hold, class 'default', on CAPI[contr1/]
--
Not all interfaces support transmitting audio before the call is answered.
It may be necessary to answer the line first, if you haven't already.
mark
On Sun, 30 Mar 2003 [EMAIL PROTECTED] wrote:
it is possible to use musiconhold. i added
exten = s,5,SetMusicOnHold,default
exten =
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