Hi
I wrote this article and at end i shared how to convert files have a look
http://younewplanet.com/index.php/articles/2012-articles-2/asterisk-configuration-step-by-step
Also i wrote an other article for file conversion you can also check that
Thank you very much Steve. Thanks for the in detail recommendations. I will
make modifications and let you know.
On Sat, May 5, 2012 at 9:22 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 5 May 2012, ABBAS SHAKEEL wrote:
I wrote this article for making the things easy
HI,
I wrote this article for making the things easy for newbies Please have a
look and let me know if you suggest some thing.
I have combined few things to gather i.e
1. Installation Steps
2. Hello World with CDR entry in database
3. Hardware configuration and loop back testing
4.
Hello
Can some body let me know any softphone that is developed using java can
support at least sip protocol. Must be open source and ready to be used. I
am trying to accomplish is to integrate it with an applet. some thing like
click to call on web page.
Sorry if this is not correct place for
Hello,
I started to work on asterisk 2 years ago. I started from book. I saved it
in google docs. You can also start from
herehttps://docs.google.com/viewer?a=vpid=explorerchrome=truesrcid=0Bxm7VSlLHvESYmZiMWYyMGUtNDI4OS00NDdjLTkwYjMtZmYxNzM0ZjQ2OGNkhl=enauthkey=CMbXtZMB
.
On Thu, Mar 24, 2011
exten = _9944NX,1,Answer()
exten = _9944NX,2,Noop(GOING FOR THE AGI)
exten = _9944NX,3,Noop(XX)
exten = _9944NX,4,Noop()
exten = _9944NX,5,AGI(//Some script here it works perfectly fine)
exten = _9944NX,6,Noop(AGI
...@gmail.com wrote:
I believe absolute timeout will do that.
http://www.voip-info.org/wiki/view/Asterisk+func+timeout
On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello all,
I am trying to end a call after a specific time period for that reason i
have tried
Hello all,
I am trying to end a call after a specific time period for that reason i
have tried various options like using S, L in the dial command. But in vain.
Now i am thinking to end the call using the AMI... but i am unable to get
the current active channel.
. i.e SIP/NT000 when i ask for
, Jan 23, 2011 at 10:14 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Sun, Jan 23, 2011 at 1:04 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello all,
I am trying to end a call after a specific time period for that reason i
have tried various options like using S, L
although I don't need the solution personally But would like to request you
that instead of posting forget it . if you post the solution to the
problem it will be more helpful.
In case some one else faces the same problem he can use your solution
Good luck
On Sun, Oct 24, 2010 at 7:10
Hello ,
Record the file and introduce echo this will give you effect of
recording and playing at same time ;)
On Thu, Jul 29, 2010 at 1:30 PM, Janu Mukherjee janu.mu...@gmail.comwrote:
Hi,
we are using Asterisk to record and playback. Both services are working
well independently but it
thing possible like that ?
On Tue, Jul 6, 2010 at 5:21 PM, Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
wrote:
On Tue, 6 Jul 2010, ABBAS SHAKEEL wrote:
Hello Community,
I have a question , I have been working with asterisk and developed some
successful
trade off..
On Wed, Jul 7, 2010 at 2:08 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-07-07 at 12:12 +0600, ABBAS SHAKEEL wrote:
Thanks to Gordon and Paul for kind help.
Actually we have a limitation to place the Asterisk server in client
premises if the server
:
On Wednesday 07 July 2010 14:58:05 Kevin P. Fleming wrote:
On 07/07/2010 10:52 AM, Tilghman Lesher wrote:
On Wednesday 07 July 2010 05:24:10 A J Stiles wrote:
On Tuesday 06 Jul 2010, ABBAS SHAKEEL wrote:
Hello Community,
. I am facing an issue of security i.e. We deploy
servers to client
Hello Community,
I have a question , I have been working with asterisk and developed some
successful applications. I am facing an issue of security i.e. We deploy
servers to client end. Now i dont want the client to see my configuration
files (Of course copy and distribute or replicate the logic
If you are using asterisk on your laptop then you dont need dahdi because
you are not going to use any hardware on your laptop.
How ever for Asterisk start at system start up you will find many
scripts
One you can do is put asterisk at end of /etc/rc.local file
On Fri, Jun 4, 2010 at 6:40
Thanks Loan Indreias ... Nice Idea
Thanks Danny Nicholas.
Cheers
On Tue, Apr 27, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:
This is probably a good idea, BUT it is likely that the dialed phone will
never ring (Perhaps that is the desired effect); In my experience it takes
() will return Dialstatus , if the number dialed is busy or off
now. use this application you can detect a number is busy or not
in several seconds. i use this method in my dialplan.
2010/4/19 ABBAS SHAKEEL shakeel.abbas@gmail.com:
Hello Community,
I Want to detect if a cell number
Thanks Motiejus Jakstsys
Thank you for the value able info i will give it a try.
2010/4/26 Motiejus Jakštys desired@gmail.com
AMI writes event Ringing..., you can catch it and (via the same AMI)
send a soft hangup request.
On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL
shakeel.abbas
Hello Community,
I Want to detect if a cell number is ON or OFF... for that matter i can
generate call to it using PSTN lines (configured with asterisk).
The problem is that i only want to see if the cell number can receive a
ring or not. If ring is recieved at called number end then mark it as
Hello Community,
I have installed Dahdi on Centos on many system and succesfully used that..
But today i have bad luck...
This is the error that i am facing
[r...@localhost dahdi-linux-complete-2.2.1+2.2.1]# make all
make -C linux all
make[1]: Entering directory
AM, ABBAS SHAKEEL wrote:
[r...@localhost dahdi-linux-complete-2.2.1+2.2.1]# make all
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-complete-2.2.1+2.2.1/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/dahdi-linux
the system... This issue has
ruined my continuous 36 hours with out sleep
One thing i have in mind is to install ubuntu instead of centos to get rid
of this issue. I may try this on Monday.
On Fri, Apr 2, 2010 at 7:45 PM, Warren Selby wcse...@selbytech.com wrote:
On Friday, April 2, 2010, ABBAS
*?
If your processor type is *i686*, you just need to install
kernel-PAE-devel-2.6.18-128.el5.i686.rpm like following:
rpm -ihv
ftp://ftp.pbone.net/mirror/ftp.centos.org/5.3/os/i386/CentOS/kernel-PAE-devel-2.6.18-128.el5
.*i686*.rpm
BR, Alexey
2010/4/2 ABBAS SHAKEEL shakeel.abbas
HEllo
try this http://www.voip-info.org/wiki/view/Digium
On Fri, Mar 26, 2010 at 3:29 PM, Faheem faheem_...@yahoo.com wrote:
Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy
X100P.
Muhammad Faheem
--
Hello,
Please Confirm if the dahdi/Zaptel service is running .
check your channels status.
On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman brama...@gmail.com wrote:
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls
are not being picked up. I don't find anything unusual in
Thanks alot for the value able information. My client is not sure about the
requirements as he reaches a final decision then i can move forward to start
working on it.
Thanks for the info.
On Fri, Mar 19, 2010 at 6:30 PM, Jonathan Addleman j...@redowl.ca wrote:
Philipp von Klitzing wrote:
I
Hello,
Please have a look to DIALSTATUS variable. here
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUSI hope it
helps
On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun bit...@gmail.com wrote:
hi,all
one problem
Hello all,
I would like to know if any one have experience with live audio streaming
like
1. Streaming from an online resource
2. Streaming from sound card AUX interface..
What i want to accomplish is that on receiving a callers call i play back a
live audio stream or stream from sound card AUX
Thanks I will look into it.
On Fri, Mar 19, 2010 at 2:26 AM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
I would like to know if any one have experience with live audio
streaming like 1. Streaming from an online resource
Look at app_ices and icecast.
hello ,
First you check out
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Once you are done with auto dial out then look for Cepstral TTS.
http://www.google.com.pk/search?hl=ensafe=activeq=asterisk+with+cepstralbtnG=Searchmeta=aq=foq=
One more thing that there are other
wrote:
Thanks guys...
I already have Cepstral installed I guess I just need to figure out
where in the .call file and format to call cepstral and then the txt
for the message. Thanks again for all of your help!
On Fri, Feb 12, 2010 at 11:50 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote
Please some one shed some light on it..
On Thu, Feb 4, 2010 at 6:48 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello All,
Please let me know Answers to the following questions .Backgroud.
1. Which one is better to use libss7 or chan_ss7. Today first time i come
to know about
Hello All,
Please let me know Answers to the following questions .Backgroud.
1. Which one is better to use libss7 or chan_ss7. Today first time i come to
know about it ... little bit i googled but need experts comment on it.
2. I have perpared my server ie installed Asterisk , Configured TE420P
why don't you post your question
On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi motamed...@gmail.com wrote:
On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com wrote:
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you
Hi,
You can directly call that class like AGI(com.abc.cde.Hello) . Hello is
class name.
Hope this helps
On Wed, Jan 6, 2010 at 2:16 PM, ahmed magdy amagdy.ibra...@gmail.comwrote:
Hello
I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started
the first example Hello AGI in
You can try this
[agi_test]
exten = 123,1,Answer();
exten = 123,n,noop(${CALLERID(num)})
exten = 123,n,set(IP_FOR_AGI=192.168.127.58)
exten =123,n,Agi(agi://${IP_FOR_AGI}/com.package.ClassName)
On Wed, Jan 6, 2010 at 2:28 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hi,
You can
Can some body shed some light on this please
On Mon, Dec 21, 2009 at 6:41 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello
when compiling asterisk with Postgresql we need to specify directory where
the postgresql is installed.
It uses some files from bin folder of postgresql (I am
to explain
Cheers
On Tue, Dec 22, 2009 at 2:21 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote:
Hello
When ever i try to use Dial DAHDI / SIP i get the following warning and
nothing happens and Asterisk moves to next
Hello
when compiling asterisk with Postgresql we need to specify directory where
the postgresql is installed.
It uses some files from bin folder of postgresql (I am not a developer of
asterisk but a user ).
I need to know once asterisk is ready to use(ie compiled and installed ). Do
it still
Hello
When ever i try to use Dial DAHDI / SIP i get the following warning and
nothing happens and Asterisk moves to next instruction
Even i know that channel is free no one else is using it
[Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable to
create channel of type 'DAHDI'
No Problem you are Welcome. But please try to be on the list
Loop back connector can be used to test your card .
For loop back connector you have to make a loopback cable. that you will
plug in card. its image is
Dear Daniel,
I am keeping this on list so that it can be help ful to others as well.
They ask me to return back to them the data packets they send me
through.
In my opinion
They are sending you data packets on some channel and you
need to reply back from another channel ie send
James you are right. let me add one more line
exten = s,n,GotoIf($[${CALLERID(num)}=]?nocid,s,1)
On Sun, Dec 6, 2009 at 3:18 PM, James Stocks stoc...@stocksy.co.uk wrote:
On 6 Dec 2009, at 08:56, Remco Barendse wrote:
I am using asterisk 1.6 at home and would like to send incoming calls
Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other key.
For this purpose i am using this
linkhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
.
*I am using this option :- *
*M(**x**)*: Executes the macro (x)
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello,
What i am trying to do is . Dail a number and ask if you wana talk to
XXX press 1 and if you dont wana talk press any other
Aah the Problem was i am working on 1.4 and in my mind and logic i was
writing code for 1.6.
The example works perfect
On Thu, Dec 3, 2009 at 3:00 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Any one have success with Dial M option, Can some one provide an example?
On Thu, Dec 3
Hello
I also tried it in begining but cant give time to it. So no success.
you can try this link
http://www.voip-info.org/wiki/view/Sphinx
http://cmusphinx.sourceforge.net/html/cmusphinx.php
http://cmusphinx.sourceforge.net/html/cmusphinx.php
hope this helps
On Tue, Dec 1, 2009 at 12:16 PM,
Hello
We need to know if a channel is not in use and can be used to dial a number
etc..
I have tried the ChanIsAvail function with different parameters.
ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc
no matter the channel is busy or not it always return 0 .
Please suggest
FYI
://issues.asterisk.org/view.php?id=14426 – link to the issue
Hope that helps.
Dan Journo
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL
*Sent:* 25 November 2009 09:59
*To:* Asterisk Users Mailing List - Non-Commercial
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Michiel and Dan
@ Michiel i have checked the variables but they dont contain any value.
@Dan I am using
.
ABBAS SHAKEEL shakeel.abbas@gmail.com wrote:
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Michiel and Dan
@ Michiel i have checked
Hello
When a user makes a call to an Asterisk system, He dials a number . We need
to know that dialed number.
We can get the dialed number by using CALLERID(dnid) and we can get the CLI
information using CALLERID(num).
I am facing problem while getting the number dialed. if the user is using
SIP
;
On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov abalas...@evaristesys.comwrote:
ABBAS SHAKEEL wrote:
I am facing problem while getting the number dialed. if the user is
using SIP phone then we can get the number dialed. but if it using PSTN
then we are unable to get the number dialed
= _.,1,Answer
exten = _.,n,NoOp(${EXTEN})
ABBAS SHAKEEL wrote:
Thanks Alex,
suppose this is the context
[abc]
exten = s,1,Answer();
exten = s,n,Noop(${EXTEN});
exten = s,n,Noop(${CALLERID(dnid)});
I get the following out put
Answer(DAHDI/2-1, )
NoOp(DAHDI/2-1, s) in new
Hello Veselin
Please try this
http://www.google.com.pk/search?hl=ensafe=activeei=PnUGS9mhLMvanAfUuozGCwsa=Xoi=spellresnum=0ct=resultcd=1ved=0CBIQBSgAq=Minimum+hardware+requirements+for+Asteriskspell=1
On Fri, Nov 20, 2009 at 3:39 PM, Veselin K vese...@campbell-lange.netwrote:
Any advise?
Hello
If we need to save CDRs on different databases for same Asterisk server ie
suppose for context [abcd] save to local:5432:abcd
and for context [wxyz] save to local:5432:wxyz
Can we manage it ? or we need to do some thing in AGI
--
Kind Regards
Shakeel Abbas
row I have to copy.
regards
Mickael
2009/11/18 ABBAS SHAKEEL shakeel.abbas@gmail.com
Hello
If we need to save CDRs on different databases for same Asterisk server
ie
suppose for context [abcd] save to local:5432:abcd
and for context [wxyz] save to local:5432:wxyz
Can we manage
Thanks Alot all. Specially Tim
It seems to be really good. I will check it in detail
On Sun, Nov 15, 2009 at 3:44 PM, Tim Panton t...@westhawk.co.uk wrote:
On 2 Nov 2009, at 12:11, ABBAS SHAKEEL wrote:
Hello,
Let me explain a scenario
There are different Asterisk Servers at different
I cant stop laughing lolz
Any how we must not reply in private but ask to post on list only. Lets
make him able to achieve his objective through the list.
Cheers
On Sat, Nov 14, 2009 at 11:56 PM, Alex Balashov
abalas...@evaristesys.comwrote:
I don't get it. I just replied helpfully to Mr.
This is happening here also :(
On Fri, Nov 13, 2009 at 9:02 PM, Cary Fitch ca...@usawide.net wrote:
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by
Hello.
I see this post many times. I have written this for you to get a start.
This is sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060; UDP Port to bind to (SIP standard port is 5060)
You may be doing some thing wrong with Configuration of Softphone. Please
take a tutorial .. Google is a good friend. I suggest you to use X-lite
softphone.
On Thu, Nov 12, 2009 at 11:25 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
Hello.
I see this post many times. I have written
I have replied you already . Please look into it
On Thu, Nov 12, 2009 at 11:31 AM, aster...@opensourcesolution.in wrote:
hi all,
i had installed asterisk on Centos 5.3, sip.conf and extentions.conf are
*vi /etc/asterisk/sip.conf*
[general]
port = 5060
bindaddr = 192.168.1.2 (asterisk
Please stay on list because if some one other face similar problem he can
get help by googling list.
IN domain name u have to specify ASterisk server IP in XLITE.
On Thu, Nov 12, 2009 at 11:36 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
I have replied you already . Please look
Aslamoalikum Ishfaq
Can you check this with asterisk 1.6.X ?
On Tue, Nov 10, 2009 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Has anyone ever had experience of phones on the same office network
being able to hear other concurrent call's audio whilst on calls of
their own? We're
Hello,
Let me explain a scenario
There are different Asterisk Servers at different Remote locations.
Recording in different formats for FIVE seconds reveals that
Format : Size
wav : 84 KB
gsm : 8.3 KB
sln : 84 KB
It can be recorded in any format. This is size for five seconds only. We
need to
convert wav to mp3 on the recording server and then send it to
the central system.
Bye,
Patrick
On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello,
Let me explain a scenario
There are different Asterisk Servers at different Remote locations
After conversion from .wav to .mp3 the size remains almost the same.
On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Patrick.
First: I dont do that intentionally.
Thanks for suggestion. Let me investigate it.
On Mon, Nov 2, 2009 at 5:34 PM, Patrick
On Fri, Oct 30, 2009 at 3:41 AM, C F shma...@gmail.com wrote:
On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
C F thankyou very much.
when i make a call to Asterisk server recieves and works fine. But as to
make external calls we have to press nine so
Hello
I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that
will be Asterisk system.
I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX
to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic
to in of Asterisk PBX).
--But i am
Thanks all
Robin Drop Box looks cool but I have developed my own code in JAVA that will
use Sockets to syncronize files across different servers.
Thanks Arjan for the link.
@ li...@torrenga.com yeah i do have considered but finally developed my own
code for sysncronization. thanks :)
if Any One
both.
On a side note, may I ask why you are integrating asterisk with the
TDA? What is the functionality you plan on gaining?
On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello
I have a scenerio to integrate an Existing Panasonic PBX with a new PBX
hello
AGI is a good option to handle such complexities
On Fri, Oct 23, 2009 at 6:33 PM, Mail list asteriskmaill...@gmail.comwrote:
Hello everyone. I have a client with specific requirement, here's the
scenario:
Call comes in
Ivr menu, press 1 for new record 2 for existing 3 for operation
at 10:08 AM, Joseph syscon...@gmail.com wrote:
On 10/20/09 17:24, ABBAS SHAKEEL wrote:
Hello
I need some advice regarding the Asterisk server that are located at
different locations.
Three asterisk servers are here each at different location. Suppose A,B,C
be
the three servers respectively
Hello
I need some advice regarding the Asterisk server that are located at
different locations.
Three asterisk servers are here each at different location. Suppose A,B,C be
the three servers respectively.
Server A is connected to server B and server C through a VPN.
I have a developed an IVR
Oct 2009, ABBAS SHAKEEL wrote:
Hello
I need some advice regarding the Asterisk server that are located at
different locations.
Three asterisk servers are here each at different location. Suppose A,B,C
be
the three servers respectively.
Server A is connected to server B and server
If you want to check in Console then NOOP can be used .if in case of
function call you can check its length if there exists some thing
On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is there any way to check if a variable is set in asterisk? I've had a
look
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
If you want
)
Ish
ABBAS SHAKEEL wrote:
Sorry its macro I called it a function.
This link will be helpful to you
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com mailto:shakeel.abbas@gmail.com
wrote
Hello
Please let me know can we call normal PSTN lines as trunk lines?? As a
normal pstn line used in home .
One More thing that If i need ten PSTN lines on one Server then which Digium
card is suitable.
I am confused with TDM800P as it say it accepts a trunk line?
--
Best Regards
Shakeel
Hello
I am thinking to develop a softphone that is integrated into web.(in form of
APPLET or some thing else)
Ie a user with with just a PC with Net Browser(fire fox etc) Installed can
make call..
Is there some thing developed before like this that is open source ??
--
Best Regards
Shakeel
Thanks.
But Can i enhance it in such away that it can make calls to asterisk as part
of a web application ??
user can call from webapplication
i think mozphone is a plugin for mozilla...
On Thu, Oct 1, 2009 at 2:00 PM, Administrator TOOTAI ad...@tootai.netwrote:
ABBAS SHAKEEL a écrit
Hello Hadi
In beginning i also face this problem . I solved it by converting to SLN
format.
You also try to convert it to sln format.
this link might help you
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
yeah it can :)
On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote:
Thank you for your reply . Excuse me , you mean the Asterisk can play SLN
files ? Can you please confirm ?
On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello
for this sip phone.
As Asterisk Server is also behind NAT. SIP phone is also in any other
network.
How can I make them communicate. As in LAN i can easily by giving asterisk
server IP.
On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
Abbas Shakeel wrote:
I Recently
Hello Hadi
While playing files extension is not specified. Remove the extension and
Enjoy
On Sat, Sep 26, 2009 at 3:13 PM, ravi kumar ravi...@gmail.com wrote:
Use
Audocity Software
Ravindra kumar
On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Can
Hello
I Recently completed an IVR application with Asterisk.
Now we are moving towards VOIP. Please give a direction how to move forward.
What i have studied so far
I am confused with NAT issues. As i can have many SIP peers on local LAN it
works but from internet it donts. We need to do
Thanks Alex
By just avoiding this will solve this problem?
On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov abalas...@evaristesys.comwrote:
Don't put a SIP server behind destination NAT. Just don't.
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System
A good way is to give try
On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
yeah it can :)
On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote:
Thank you for your reply . Excuse me , you mean the Asterisk can play SLN
files ? Can you
abalas...@evaristesys.com wrote:
Don't put a SIP server behind destination NAT. Just don't.
Why not? Mind to explain?
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System that is placed some where on local LAN (suppose in
office A) A sip(or any other
Hello all
Is it possible for Asterisk (it can be Asterisk Manager) to
A caller who is flowing in any dail plan (Say Dialplan A) ...On a particular
event ( can be generated by any other caller) pick him up to any other dial
plan(say Dial plan B) without his(/her) wish. .. and after taking some
, 2009 at 3:23 AM, Christian Victor
christ...@victormedia.dewrote:
2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com
Thanks you very much Kevin.I will try it by connecting one end of
Ethernet cable to one slot and other to second slot . Configuring one
as pri_net and the other as pri_cpe
...@digium.comwrote:
ABBAS SHAKEEL wrote:
But I cant generate calls using the loop back connector and get the
following error
*CLI [Sep 9 14:42:55] NOTICE[9981]: channel.c:3749
__ast_request_and_dial: Unable to request channel DAHDI/1/123
You cannot use a loopback connector for a PRI
HelloI have the following system
Asterisk 1.6.1dahdi 2.2.0.2
TE420P card
Centos
I have noticed that all the four lights are blinking(ie coming red and then
off so on)...
Previously I also noted that when dahdi drivers are not installed lights
blink but one by one in sequence(like in marriage
cant generate calls using the loop back connector and get the
following error
*CLI [Sep 9 14:42:55] NOTICE[9981]: channel.c:3749 __ast_request_and_dial:
Unable to request channel DAHDI/1/123
THANKS
On Wed, Sep 9, 2009 at 12:47 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
HelloI have
Hello All
I am facing a strange exception/Notice while running an Asterisk box.
Let me explain in detail
I have asterisk box 1.6.1.2
centos 5.2
I have connected TE420P to the box. then connected a loopback connector(
http://i580.photobucket.com/albums/ss246/shakeelabbas/loop_back.jpg). The
/etc/dahdi/system.conf file is auto generated do we need to change in this
file as we do for zaptel ?
Any working examples
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
right?
*De:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *En nombre de *ABBAS SHAKEEL
*Enviado el:* Lunes, 07 de Septiembre de 2009 11:33 p.m.
*Para:* Asterisk Users Mailing List - Non-Commercial Discussion
*Asunto:* Re: [asterisk-users] E1 line
Hello
Dial(Zap/3/5551234)
here 3 is the channel. 5551234 is PSTN number
how ever you will have a better understanding after reading this
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
2009/9/7 Songtao Yu yustao_2...@hotmail.com
Hi All,
I am new to Asterisk. Now I got one question on
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