You probably have a cron job running that executes 'asterisk -rx'
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati
Sent: Monday, April 05, 2010 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, Mar 30, 2010 at 4:16 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello All
do you have ant software in order to change the format from mp3 or wav to
gsm in order to using it in asterisk file
thank you so much for your help and support
Best Regards,
salah
If you
On Mon, Mar 29, 2010 at 2:26 PM, Mike Diehl mdi...@diehlnet.com wrote:
On Monday 29 March 2010 10:15:50 am jon pounder wrote:
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive
Hi All
I'm involved in discussions with my carrier right now and am wondering if
anyone has interconnected Asterisk to Metasphere via SIP?
Thanks
Dave
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On Wed, Mar 17, 2010 at 10:16 AM, Daniel Leite de Abreu
dlab...@gmail.com wrote:
-bash-3.2# cd dahdi-linux-complete-2.2.1+2.2.1/
-bash-3.2# make all
make -C linux all
make[1]: Entering directory
`/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux'
make -C drivers/dahdi/firmware
On Wed, Mar 17, 2010 at 5:40 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
Hello,
I was wondering if the following was possible:
When somebody sends a fax to my direct number 0101234567105 (my extension
will be 105) is it possible that Asterisk, or an addon sees this as a fax,
and
On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena moh...@starcomms.com wrote:
I have been trying to do this since 2 days but couldn't make itneed your
help..
Well, you could certainly ask Cisco for help.
You did pay Cisco money, right?
PSTN-Cisco AS5350---Asterisk BoxVoIP
On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
from my mobile which comes into our server and answering the number on a
particular SIP extension which all works fine. I'm then dialling an
snip
Bumping a thread without adding anything useful is annoying. If you do
it again, I won't be helping.
/snip
Although I have gotten quite a chuckle from your posts, it's really going to
hurt when you fall from that high horse.
--
snip
and also to do LCR and Quality based routing of International calls?
I don't know what that means.
/snip
LCR = Least Cost Routing
Routing a call based on the quality or cost of a route (PSTN term vs SIP to
PSTN term vs SIP to SIP) is actually quite common.
--
On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
David Backeberg wrote:
On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
You didn't mention version. Could be relevant.
Apologies for not adding the version, it's 1.4.17
Yeah, that's relevant
On Fri, Mar 5, 2010 at 3:06 PM, Robert McGilvray rmcgi...@globeop.com wrote:
Does anyone use confbridge in a large installation and can provide feedback
on its stability, quality in comparison to MeetMe? I use a sangoma card in
my 1.4.2 box to provide timing and it has never been an issue. Can
I would love to see any info on this as well. I see similar issues with meetme
bridges having locked channels. It's easy to set a timeout but a fix (maybe I'm
just doing something wrong?) would be better than a workaround.
-d
-Original Message-
From:
On Mon, Mar 8, 2010 at 1:42 PM, Franklin Webb fw...@imminc.com wrote:
Hello David,
I had an application where I had to pass data between Asterisk and a Genesys
system using SIPAddHeader(). It worked pretty well, but we had two genesys
boxes, and by the time I was done I found I was losing
1.6.0.25
Asterisk-addons 1.6.0.4
Libpri 1.4.10.2
I have install libpri first and then asterisk.
Regards
David.
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David Little
MM Technology, Inc.
da...@mandm-tech.com
704.882.9432 x3
704.882.0405 FAX
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New to Asterisk? Join
/lib/asterisk/sounds/30-minutes-of-silence.gsm ;*
*
*
*
*
*
*
On Fri, Mar 5, 2010 at 4:36 AM, David @ULC ucoms2...@gmail.com wrote:
I believe we GSM of 8 bit for Asterisk ?
On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote:
Record a muted channel for 30 minutes like
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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, 2010 at 4:21 AM, David @ULC ucoms2...@gmail.com wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
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I believe we GSM of 8 bit for Asterisk ?
On Fri, Mar 5, 2010 at 4:35 AM, David @ULC ucoms2...@gmail.com wrote:
Record a muted channel for 30 minutes like this:
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,record(silence_long.gsm|1800|s)
exten = s,n,hangup
Greetings:
I'm in the situation where I'm trying to splash information picked off
by an asterisk IVR into a Cisco call center environment. I'm under the
impression that the ONLY way to do this is to setup socket connections
with the Cisco voice processor, or CVP, and send packets
corresponding to
On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs szasz.szabo...@gmail.com wrote:
Hi,
I have encountered a DTMF issue. My scenario:
Access carrier-sip
Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM Switch
the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk
On Mon, Mar 1, 2010 at 9:40 AM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 1, 2010 at 9:25 AM, Szasz Szabolcs szasz.szabo...@gmail.com
wrote:
Hi,
I have encountered a DTMF issue. My scenario:
Access carrier-sip
Asterisk-1.4.25.1-sipCiscoGW-ISDN-TDM
On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote:
I am trying to get the usernum of a user when dialing in to a MeetMe
conference. Is there somehow a possibility to save the usernum of a
MeetMe participant into a variable? Everything should be done through
the DialPlan, no manager
On Mon, Mar 1, 2010 at 11:01 AM, Emrah e...@ekanet.net wrote:
Hi!
Thanks a lot for your answer.
The problem with the command you mentioned is... When do I call it? If two
people happen to enter the conf at the sametime,
I have a feeling there may be some little confusion there...
Do you
On Fri, Feb 26, 2010 at 11:24 AM, Jay Vocaire jvoca...@innproc.com wrote:
I am new to Asterisk and have searched all over for an answer to this,
so please don't skewer me too bad if this is a stupid question. I am
currently running 1.6.0.21 on a few test boxes (one i386, one x64), and
have
snip
Does anybody use the Morsecode app for anything interesting? I'm strangely
fascinated by this core piece of Asterisk functionality.
/snip
Duh! How are we going to spread the word about how to take those alien bastards
down if we don't keep morse code around!?!??!
On Wed, Feb 24, 2010 at 9:20 AM, Juan Sandro juan.san...@hotmail.com wrote:
Hi Guys
We are using asterisk 1.4 on all of our platforms for a while now.
Some of our partners recommended to use asterisk 1.6 in order to improve
overall stability and performance.
Can someone please let me know
Have you check if MySql is already running?
Have you check HD space?
regards.
2010/2/24 ahmed magdy amagdy.ibra...@gmail.com
Hello,
Asterisk Real time database worked on astersik 1.6.2.0 but now i am working
on Asterisk to latest version which is 1.6.2.2 ,there is a a warning
[Feb 24
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1,
sip-silence) in new stack
-- Playing 'sip-silence' (language 'en')
-- Executing AGI(Local/91441425477...@default-b9f2,1, agi://
127.0.0.1:4577/call_log) in new stack
I changed my VOIP, and now things are ok.
But didnt understand, how can VOIP can affect it ?
On Wed, Feb 24, 2010 at 11:53 PM, David @ULC ucoms2...@gmail.com wrote:
*Code:*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback(Local/91441425477...@default-b9f2,1
On Mon, Feb 22, 2010 at 8:20 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Hi,
Does anybody have any experience with asterisk where are four PCIe cards are
used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
I would recommend calling
On Sun, Feb 21, 2010 at 10:04 PM, Sean Brady sbr...@gtfservices.com wrote:
I do get choppy audio when playing recordings occasionally. I haven’t had
time to figure that one out, but I haven’t put it into production yet.
You just said you're getting unexplained choppiness.
You also just said
On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman j...@redowl.ca wrote:
David Backeberg wrote:
Timers are built on the premise that they have access to either a real
timing device, or unobstructed access to a processor which clicks
through a proc cycle at a pre-determined rate. Once you
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the security and feature
improvements over the
I think Vicidial, works great.
Regards.
2010/2/22 Apa Minerala apaminer...@yahoo.com
Hello,
We used to recommend a commercial software but client is a small callcenter
who cannot afford something big.
Would you recommend something open-source which could work for a 40-seater?
Thank
Its crazy
I made it working .
Today I had to reinstall all due to soem reason.
Now, when I am trying, its NOT coming.
Same CPU, Same Lan, Same Windows which acts as Internet Gateway.
CALL Doesnt hit my Asterisk.
http://i50.tinypic.com/1z3axrc.jpg
http://i45.tinypic.com/23mr5uq.jpg
--
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti
mleone...@evolutionce.com wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote:
Hello David,
Thanks so much for your message!
Please check my comments inline below...
David Backeberg wrote:
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm trying to figure out how
On Wed, Feb 10, 2010 at 2:13 PM, Leo Burd l...@media.mit.edu wrote:
How much control do the ssh processes have over the call, if any?
It occurred to me that I might be answering this backwards.
So from the perspective of server A, trying to talk to a remote system
B running asterisk, server A
I dont have a Static IP.
How can I ask IPKall to send call to my Asterisk ?
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???
On Thu, Feb 18, 2010 at 2:45 AM, David @ULC ucoms2...@gmail.com wrote:
I dont have a Static IP.
How can I ask IPKall to send call to my Asterisk ?
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Looks like IdeaSip need STATIC ip else it doesnt work.
.
On Thu, Feb 18, 2010 at 3:02 AM, David @ULC ucoms2...@gmail.com wrote:
Ok
I can use
Dyndns.org
I registered myself.
easy.selfip.com
https://www.dyndns.com/account/services/hosts/easydialnow.selfip.com
successfully activated
for '11012012...@66.54.140.46' timed out, trying again (Attempt
#18)
When I try to Ping from my CentOS , I can ping 66.54.140.46.
On Thu, Feb 18, 2010 at 3:11 AM, David @ULC ucoms2...@gmail.com wrote:
Looks like IdeaSip need STATIC ip else it doesnt work.
.
On Thu, Feb 18, 2010 at 3:02 AM, David
Feb 17 19:19:04 NOTICE[2554]: chan_sip.c:10077 handle_response_peerpoke:
Peer '11012012600' is now TOO LAGGED! (2567ms / 2000ms)
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC ucoms2...@gmail.com wrote:
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings
proxy.ideasip.com, port 5060
-- Got SIP response 479 Please don't use private IP addresses back
from 208.97.25.11
On Thu, Feb 18, 2010 at 5:34 AM, David @ULC ucoms2...@gmail.com wrote:
hmmm Ok..
Is this a Asterisk Question ?
I have a setting as :
Global Settings
http://i50.tinypic.com/120rwya.jpg
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC ucoms2...@gmail.com wrote:
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure
208.97.25.11
I cant use Ideasip ???
On Thu, Feb 18, 2010 at 7:12 AM, David @ULC ucoms2...@gmail.com wrote:
So, this will change :
register = 11012012600:passw...@proxy.ideasip.com/11012012600
[ideasip]
type=friend
secret=password
username=11012012600
host=proxy.ideasip.com
insecure
On Mon, Feb 15, 2010 at 11:05 AM, Amit Patkar | Avhan Technologies
Pvt. Ltd. a...@avhan.com wrote:
I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for
PSTN-IP gateway. What is the maximum call handling capacity I can achieve
with this server?
You can handle a lot of
I use IdeaSip with IPKall.
How may channels are open when we use IdeaSip ?
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On Sat, Feb 13, 2010 at 10:50 PM, Alejandro Recarey
alexreca...@gmail.com wrote:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command?
On Thu, Feb 4, 2010 at 7:39 PM, Lyle Underwood lyleunderw...@gmail.com wrote:
If call recordings were stored in stereo and the callers were evenly
distributed along the stereo spectrum. BAM.
Cisco has this. It's called telepresence. It costs a LOT of money, and
takes a LOT of bandwidth, but you
On Tue, Feb 9, 2010 at 6:24 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
I wonder what mute should mean. Does it mean that the participant will
not receive any media, or that media sent by the participant will be
ignored, or both?
Please post your discoveries to:
On Tue, Feb 9, 2010 at 10:26 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Answering myself: muting means that the participants voice is ignored.
Thank you for updating the wiki and the list.
I looked into this when I was having problems with early 1.6.0.*
MeetMe(), specifically the
Hi,
Using asterisk 1.6.2.0 I have a queue definition with strategy=linear.
How do I jump to the next dialplan item after having tried
(unsuccessfully) all queue members?
If I use Queue(test,n) then only the first member is contacted. And if I
omit the n option then all members are retried
On Sun, Feb 7, 2010 at 9:54 PM, Leo Burd l...@media.mit.edu wrote:
Hello there,
I'm trying to figure out how to run a PHP script on a remote machine and
still have access to the audio stream associated with the call.
Ideally, I'd love to play/record audio files directly from/to the remote
(better ways)? Which of the methods would be fastest for a
large number of endpoints, and most reliable? Any help is greatly appreciated.
With kind regards,
David de Boer
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I've been waiting for this for years. Except that snom phones are crap -- I
would really like to see openvpn or ssh tunneling hacked into a Cisco phone...
But it's still awesome.
-dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
snip
I have upgraded the phones to the most recent firmware (POS3-08-11-00)
and * is Version: 1:1.4.21.2~dfsg-3+lenny1 (debian).
snip
That doesn't look like cisco firmware to me... Unless I'm mistaken. What
version are the phones on? (Settings = Status = Firmware Versions)
-Dave
--
On Mon, Feb 1, 2010 at 8:55 AM, Peter den Hartog
peterdenhar...@gmail.com wrote:
I'm using the default Asterisk function Monitor to record calls, but i have
some issue's with this, the problem is when a call is finished, it never mix
in out together, bellow you can see my call configuration:
On Fri, Jan 29, 2010 at 3:09 PM, Hristo Benev hris...@smartbox.ca wrote:
If I use call file with spool
Fax is send but if I use manager
I get
Any suggestions?
Well, one obvious solution is to just use call file. Problem solved.
Try changing your call manager setup to use a Local channel
snip
However, if you're going to be doing
massive joins for reporting, you're better off using something else (or
running individual MySQL slaves, whose purpose is to run those complex queries
and doing nothing else). In a past life, our MySQL database ran circles
around Oracle, Informix, and
This is WAY OT but I had no idea what fnal.gov was, so I checked it out:
http://computing.fnal.gov/xms/Services/Getting_Services/Web_at_Fermilab/Professional_Home_Pages_at_Fermilab
And I quote ...professional information about themself...
About themself? Really? Really?
That is all.
Cheers
snip
many people around think mysql is not a good option for database, they
think mysql
is only suit for small business. but i want to have a try. i need to
convince them to use this.
/snip
This statement is absolute BS. Give me some factual, backed statements by
trained database professionals
On Sun, Jan 24, 2010 at 2:51 AM, Magnus Benngård
magnu...@inputinterior.se wrote:
Morning,
Have some questions regarding receiving and sending faxes...
1:st example:
exten = 101,1,Answer()
exten = 101,2,Wait(3)
exten = 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff)
You example is
Admittedly I didn't read your SIP debug (on the mobile), but do you have
reinvite=no set for the extensions and SIP trunks (providers)?
This sounds on the surface like a classic case of the Mondays. Erm reinvites I
mean.
snip
1. Incoming call from pstn/viop provider
2. Call is answered by a
Some rack-mount servers I've encountered have an option to have the
older-style PCI slots available in at least some slots. If you're
really just using four FXS/FXO ports, it's unlikely you need very much
horsepower, and you could use an older system for the foreseeable
future.
If you really need
On Fri, Jan 15, 2010 at 1:54 AM, randall rand...@songshu.org wrote:
does anybody know of another solution to this or is my conclusion above
simply all the choice there is?
So let me get this straight.
You're planning on buying multiple Gigabit, PoE switches, and you're
quibbling over the price
On Fri, Jan 15, 2010 at 12:27 PM, Bruce Nik brucev...@gmail.com wrote:
Hi Guys,
Other than than yum repository (which fails when installing freepbx with it)
are there any automated install scripts out there that would install
Asterisk 1.6 or 1.4 onto a CentOS LAMP system?
If the script
On Fri, Jan 15, 2010 at 3:15 PM, Bruce Nik brucev...@gmail.com wrote:
Provided there is no comprehensive install guides (or is there?) yes I would
like to see an easy install script which can install it all.
tar xvzf
./configure
make
(optional, do a 'make menuconfig')
make install
But the
Are you actually trying to strip off the + or are you doing it as part of
trying to check the callerid number to see if it is valid. If the later,
then consider REGEX()... here is a snippet from my privacy manager script...
; First lookup number in asterisk DB for a Caller ID name. exten =
snip
But then the other peer says:
-- Called *31#w06123456...@xs4all-out
-- SIP/xs4all-out-0234 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/evert-0233' status is 'CONGESTION'
Anyone an idea where i should look, or
snip
This doesn't work?
Dial(SIP/*31#ww061234123412)
/snip
When I was browsing the sip debugs, it seemed that the 'w' was not being
honored for one reason or another. My thought at the time was maybe it didn't
work at all over SIP.
Does the w *just* work with dahdi or does it work over sip as
snip
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
/snip
Thanks Kevin, that's what I figured (though not quite so concisely)...
Going foward, is there any way to programmatically
snip
Going foward, is there any way to programmatically inject DTMF tones into an
already-bridged channel?
/snip
Well, due to the lack of responses, either I missed something obvious or nobody
cares. I'm really hoping I didn't miss something obvious... :).
In any event, I got curious of my own
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US toll-free: +1 888 842 2720
UK: +44 (0
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On Fri, Jan 8, 2010 at 8:59 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
This is what I was using at the time:
asterisk-1.4.21.2
I really, really prefer the faxing in 1.6. It's so nice to configure
compared to 1.4. I'll leave it to the ChangeLog and anybody else
I would have read your message but I couldn't find it amongst all of this
garbage...
:)
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Friday, January 08, 2010 11:10 AM
To: Asterisk
And how will we ever re-write the 10+-year-old RFCs which no longer hold
relevance to modern email clients if nobody goes against the grain and does
what makes sense rather than what has been generally accepted?
-Dave
snip
And to add on to this: aside from whether you think it is silly or not,
On Fri, Jan 8, 2010 at 1:47 PM, Daniel Araujo redsna...@gmail.com wrote:
I have the same issue on my Asterisk installation (Asterisk 1.4.25).
As you can see, the T38 module isn't enabled on my installation. Tried ask
google how to make it work, but found no hints yet.
Anyone can help us?
If
Gmail DOES process those headers...
And a proper mail client will also parse the headers and provide unsubscribe
information/buttons based on that
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I haven't had a good mailing list war in a while.
Yes, gmail DOES default to top posting, because bottom posting is silly (in
general, but especially for a client that hides quoted text (like gmail)). Top
posting is modern. And better. And doesn't make me scroll through 10 thousand
messages
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. I would like a timeout of 1 second, then return.
A few things...
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the same
ReceiveFAX application :
2. sendfax/hylafax/iaxmodem asterisk spandsp
In the 2nd case, I've got 3 craches out of 3 attempts (with a rough
On Mon, Jan 4, 2010 at 6:44 PM, Karl Fife karlf...@gmail.com wrote:
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x
instance AND do it reliably? If so, I can think of a number of locations
with copper loops that could be scrapped. I'm actually quite surprised at
what
2009/12/29 Vinícius Fontes vinic...@canall.com.br:
Hello everyone.
I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing.
Voice is working great, but I never configured anything using T.38 in
Asterisk so I'm kinda lost.
So you're trying to use a SPA8000 to act as a
On Wed, Dec 23, 2009 at 4:21 PM, Sascha Ferley
sascha.fer...@infineon.net wrote:
Hi,
I am in need of ordering a new server here for our asterisk solution. Since
the corporate standard is Dell we need to stick to a dell server. We used to
deploy 2900III without any issues, however now they are
CDR record!
== Spawn extension (tutorial, 4321, 1) exited non-zero on
'SIP/ivan-0a07dc80'
it says Failed to record Radius CDR record. Could you tell me ,
what's wrong with it?
2009/12/23 Olle E. Johansson o...@edvina.net:
23 dec 2009 kl. 11.25 skrev David Cunningham:
Shukun
, David Cunningham
dcunning...@voisonics.com wrote:
AsteriskWin32 does have SIP server functionality, same as the linux
version.
I can't think of any reason why having your CentOS Asterisk be both client
and server and register with itself wouldn't work.
Although I am wondering how much
/asterisk-users
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Voisonics
IVR development, VOIP consultancy
http://voisonics.com/
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to another SIP provider...
On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote:
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hadi,
You could use Asterisk as a sip server, it's installable on Windows.
Using sip set debug on might
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, every 5 minutes
exten =
s,n,Dial(${AVAILCHAN_NOSESSION}/${ARG2}#,,rL(360:300:30))
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*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Juan David Diaz
*Sent:* Tuesday, 22 December 2009 11:40
.
Brian
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Good Day List Users,
Is there any way to play an audiofile or at least a beep into an established
call, I want to do this event each 3 minutes in the call, for now I have a
shell to get the call time and evaluate the 3 minutes.do you know any
way to play that sound?
I tried app_inject, it
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