You're a genius, sir! I don't know how I missed the part about ports, but
anyway...
Looks for "channelvars": {
"UNICASTRTP_LOCAL_PORT": "*14880*",
and then
vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
'#transcode{vcodec=none,acodec=*a*
On Sun, Jan 3, 2021 at 4:14 PM Jonathan H wrote:
> Very simply, I want to pipe some external audio into a channel (bridge)
> using the externalMedia channel option.
> Running Asterisk 18 on ubuntu, here's what I did to try and test things
> out:
>
> open a console tab
> vlc -vvv
Very simply, I want to pipe some external audio into a channel (bridge)
using the externalMedia channel option.
Running Asterisk 18 on ubuntu, here's what I did to try and test things out:
open a console tab
vlc -vvv https://media-ssl.musicradio.com/LBCUK --sout
Hi All,
I have installed centos 5.6 32 bit on xeon server and i have also installed
latest version of asterisk 1.6 and dahdi as well.
I want to install chan_ss7 for this server and I want to know about the
following device.
Digium TE420B
I dont know much about the configuration files for Digium
...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Help needed for chan_ss7 for Digium device
Hi All,
I have installed centos 5.6 32 bit on xeon server and i have also installed
latest version of asterisk 1.6 and dahdi as well.
I want to install chan_ss7 for this server and I want
Hello Asterisk List,
My company is running a bunch of Asterisk servers behind a Kamailio
(openSER) SIP proxy gateway. Calls come in from our PTSN to VOIP
service to Kamailio, which then randomly chooses an Asterisk server to
handle the call. All Asterisk servers are 1.6.0.9, but the issue
Chris Kairalla wrote:
This is great, and asterisk will deny calls once it hits this max
number. OK, here's where I get stuck. Once that max is reached,
Asterisk will send a 480 Temporarily Unavailable, which is great,
and a BYE which is not great, as that gets passed along to the PTSN
Hi, I require some urgent advice/help concerning chanspy.
We have a web interface that lists all bridged SIP calls (show channels
concise) When we click on the call, our local extension rings. When we pick
up to spy, parties involved in the call that we are spying on are suddenly
unable to hear
Hi, all
Can someone give me an example on how to do following:
Asterisk receives incoming call from SIP
Asterisk asks for a pin number
Astersisk provides dialtone
Asterisk collects digits from the caller and places a call on another interface
Any pointers are greatly appreciated.
Thanks,
The DISA application should do what you are looking for.
PaulH
Rudolf Ladyzhenskii wrote:
Hi, all
Can someone give me an example on how to do following:
Asterisk receives incoming call from SIP
Asterisk asks for a pin number
Astersisk provides dialtone
Asterisk collects digits from the
I have this in my extension.conf:
[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Wait(1)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include = fax2emailstart
[fax2emailstart]
exten =
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say fmt
Thanks!
David
___
--Bandwidth and Colocation
David Thomas wrote:
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not say
fmt
Your wav files are broken.
(or maybe in a format not supported
On Wednesday 17 October 2007 08:18:01 David Thomas wrote:
I get the following error when trying to play wav files for my IVR
menu. Does anyone know what this means or how to fix it?
[Oct 17 01:04:23] WARNING[9799]: format_wav.c:124 check_header: Does not
say fmt
It means that your .wav files
We've just received a bill from bt where it claims that we are making
numerous calls to the same number time after time.
e.g.
01226xx Barnsley20/06/2007 211516:00:00
01226xx Barnsley20/06/2007 121908:55:32
01226xx Barnsley
Telco Switching could be waiting for a ten digit number. I know that
Sprint and some others expect ten digit local calls
On 2/28/07, Mark Engelhardt [EMAIL PROTECTED] wrote:
Hello,
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I
Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never
had to rerun that when I made changes in that file before, but we can try
it.
MY WISH: TelCo Switchmen could talk intelligently about the protocols used
on PRIs!
On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Matt
You sure that you have local service on the PRI? Maybe it's just an LD PRI?
On 3/1/07, Matt [EMAIL PROTECTED] wrote:
Yes asterisk was stopped and restarted. ztcfg was not rerun. I've never
had to rerun that when I made changes in that file before, but we can try
it.
MY WISH: TelCo
Hello,
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I believe code 28 means Invalid
Number) and I hear a fast busy on the phone.
Here is the output:
-- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack
-- Requested
Hello,
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I believe code 28 means Invalid
Number) and I hear a fast busy on the phone.
Here is the output:
-- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack
-- Requested
On Wed, 28 Feb 2007 19:13:55 -0500, Mark Engelhardt [EMAIL PROTECTED] wrote:
Hello,
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I believe code 28 means Invalid
Number) and I hear a fast busy on the phone.
Here is the output:
Mark Engelhardt wrote:
I just installed a PRI and when I make a local (seven digit) call, I
get Code 28 back from the telco, (I believe code 28 means Invalid
Number) and I hear a fast busy on the phone.
Here is the output:
-- Executing Dial(SIP/marke-17b1, ZAP/G1/4967171) in new stack
right and the asterisk debug is showing it going out as 7 digits whch
the telco says is the way local should be dialed but yet the telco is
seeing extra zeros on the end. we already know the ton is
wrong...the question is where are the extra digits coming from.
On 2/28/07, Nic Bellamy [EMAIL
Matt wrote:
right and the asterisk debug is showing it going out as 7 digits whch
the telco says is the way local should be dialed but yet the telco is
seeing extra zeros on the end. we already know the ton is
wrong...the question is where are the extra digits coming from.
localprefix or
So the questions: Is there anyway to further verify that
asterisk is
not sending any extra digits or filler digits to the telco on
the PRI? If the problem is not in asterisk or zaptel, what do
I say to the
Telco to get them to believe the problem is on their end?
At the console
Matt wrote:
right and the asterisk debug is showing it going out as 7 digits whch
the telco says is the way local should be dialed but yet the telco is
seeing extra zeros on the end. we already know the ton is
wrong...the question is where are the extra digits coming from.
Did you try unknown
It is currently set to unknown.
switchtype=national
signalling=pri_cpe
pridialplan=unknown
On 2/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Matt wrote:
right and the asterisk debug is showing it going out as 7 digits whch
the telco says is the way local should be dialed but yet the telco
Matt wrote:
It is currently set to unknown.
switchtype=national
signalling=pri_cpe
pridialplan=unknown
Was it originally or did you just change it? Did you stop Asterisk and
do ztcfg after making the changes?
Thanks,
Steve
___
--Bandwidth and
That has been the setting all along.
Matt and I will try pri intense debug span 1 in the morning and we
can go from there.
Thanks so much for your help so far.
I am also trying to get the telco to tell us what they are actually
seeing on their side, The report of 000 at the end of the
Folks,
How much efforts are needed to make Asterisk code to run on Vxworks?
Is there any document in the distribution which describes the steps to
follow to run on Vxworks.
Is there any limitation in Vxworks which should be disabled or remove
in Asterisk server code.
I'm using Polycom Soundpoint phones and I want to use some extensions beginning
with # for features setup. I'm getting the fast busy can't match it signal. I
want to match #50 for call forwarding, for instance, and #505551212 to set the
call forwarding number and turn it on. I have tftp set up
John French wrote:
I'm using Polycom Soundpoint phones and I want to use some extensions beginning with #
for features setup. I'm getting the fast busy can't match it signal. I want
to match #50 for call forwarding, for instance, and #505551212 to set the call forwarding
number and turn it
Hello all.
I have two identically configured asterisk servers each with a TDM2422P
(with two S400M FXS modules and two X400M FXO modules).
*1 works perfectly and the sound quality is great. However, I am having
audio quality problem with *2 when making or receiving calls over the PSTN -
the
Hello there,
For some reason, Asterisk is crashing everytime that my system tries to do
anything related to ODBC. I would really appreciate if anyone could give me
ideas or pointers to the solution of this issue...
Here's what I've found out so far:
* I can run isql -v my_dsn username
hello all,I have just installed [EMAIL PROTECTED] version 2.1. It keeps working fine for couple of days. But after couple of days I start getting the followingerror as the Asterisk does not start automatically so I try to start it with "asterisk -vc". Any ideas? and how to fix this error.
On Tuesday 10 January 2006 11:40, Amir Aziz wrote:
I have just installed [EMAIL PROTECTED] version 2.1. It keeps working fine
for couple of days. But after couple of days I start getting the following
error as the Asterisk does not start automatically so I try to start it
with asterisk
Good Day list,
I have read wiki pages I have googled to death and am getting no
closer to understanding the methodology of onhold music.
Maybe I am trying to do something that is just not possible:
Here is my desire.
1) Call comes in to the asterisk box via Zap
Hi All,
I've got Asterisk working and am trying to configure with Sipgate. I can make out going calls. Incoming calls show up on the AMP panel with the trunk showing red. However, the call does not go to the extension.
I initally configured Asterisk by editing the config files. I have followed the
Hi,
Could I get the complete configuration on SMS part? The files I have
to modify and the messaging center I should do the modifications the extra
look outs I should do?
Thanks,
H.Gireesh
USTechnology
Gireesh Hariharasubramani
[EMAIL PROTECTED]
Bhavani,
Ground Floor,
Could I get the complete configuration on SMS part? The files I have
to modify and the messaging center I should do the modifications the extra
look outs I should do?
Check the wiki for a fairly extensive page on the SMS app:
Hi,
I am trying to change the dialplan to
enable call recording (incoming and outgoing calls) on the click of a
button. Is it possible? All the documentation I found so far, enable
recording for all calls to an extension.
Does this code look ok?
Currently Recording
on only for 1030
I think you're looking at this the wrong way. Take a look at automon in
features.conf. Play the for-quality-purposes disclaimer/misleader on
all incoming calls to these extensions and use the w option on Dial().
What you've done below won't record an existing call.
Mark
Swapna Gupta wrote:
Hi.
I have the following line in the default context of all my internal extensions:
exten = 9876,1,Transfer(125)
When I dial extension 9876 from any sip phone, * dutifully transferrs it to extension 125, which is just what I want.
Unfortunately when I dial 9786 from my Zap connected analogue
Why don't you just use Dial(SIP/125)??
Or better, if you have your extensions defined in context e.g.
[from-internal], just do:
exten = 9876,1,Goto(from-internal,125,1)
Julian.
On 7/8/05, Mark Edwards [EMAIL PROTECTED] wrote:
Hi.
I have the following line in the default context of all
... this will get you thru.
Regards:
Bharat M. Sarvan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sharath
Chandra
Sent: Friday, May 13, 2005 5:40 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Help needed on setting up realtime
I
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call
in and through successfully. I was trying to set up the Realtime -
picking the sip.conf and extensions.conf from mysql. I was going
through some wiki pages, but what i don't understand is - which
configuration change makes
hello
i need help on PSTN Calls via quintum gateway. i have
a simple problem when i am try to send INVITE to PSTN
quintum gw. it is replying me 183 session progress and
call duration is starting at this point. after this he
is sending ringing then 200 OK. Billseconds are
incorrect in this case.
Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card and now
want to setup my dial plan. With some help from the suggestions given online
I have been able to configure the two SIP phones to interact with each
other.
I want to use this to call on to a Telecom line(PSTN) and vice
Sent: Thursday, May 05, 2005 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help needed with PSTN line
Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card and now
want to setup my dial plan. With some help from the suggestions given online
I have been
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Salina Jain
Sent: Thursday, May 05, 2005 10:23 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help needed with PSTN line
Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card
U don't need to have sound device for * sound service running
just make sure that you have in modules.conf
noload = chan_alsa.so
noload = chan_oss.so
On Wed, 2005-04-20 at 08:44, [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk-1-0-7 and running it succesfully. But iam unable to use
the
List,
Is there someone on the list that is using or testing the F1000 handset of
Utstarcom?
If so, can you help me in setting it up on an asterisk box? I will use it as
a wire-less phone when I am at home, and similar as a cell-phone when in
reach of free AP's like in the airports etc.
What
Hi,
I installed asterisk-1-0-7 and running it succesfully. But iam unable to use
the sound services.
I have the following warning messages when i launch asterisk
Apr 19 21:15:40 WARNING[10918]: chan_oss.c:992 load_module: XXX I don't work
right with non-full duplex sound cards XXX
==
Hi Everyone,
I have bought a Digium
TDM400P and am using asterisk on RedHat 9.0. I was able to configure Asterisk
and SJPhone. When I call from the softphone to the asterisk it registers and
then give me a call back on the SJPhone and then shows both are connected and
whenever I call
Hello all,
I Have to install an asterisk based PBX on a large Bussines, about 200 extensions, where the phone is a very critical service, this bussines need to be called and call the whole day.
I am thinking to install two asterisk servers with the same config, and if one of them will be broken
On 7 Mar 2005, at 12:10, [EMAIL PROTECTED] wrote:
Hello all,
I Have to install an asterisk based PBX on a large Bussines, about 200 extensions, where the phone is a very critical service, this bussines need to be called and call the whole day.
I am thinking to install two asterisk servers
Ismael,
I'm not going to give you a full answer, because this is a big topic,
and I sell high availability systems to my own customers. Having said
that, here are some ideas. This list is not definitive, and I'm sure
other people will have other suggestions.
You have 2 issues:
1. Keeping the
On 12:41, Mon 07 Mar 05, Alistair Cunningham wrote:
snip
If neither of the above are possible, consider using IP takeover. This
is a tricky thing to make work 100% - you may need expert help.
I got this up and running with CARP in half a day. Learned
PF, OpenBSD installation and CARP setup.
I need some help with installing Asterisk with ZAPHFC on my Linux system. I've
installed Red Hat 9 with kernal 2.4.20-8 and upgraded it to 2.4.31-9.
I've donwloaded bristuff (several versions) but all fail to compile. There
seems to be some includes missing in /usr/includes/linux. Although I
CAPS LOCK fUnNy
On Sun, 2005-02-27 at 20:25, Roy Sigurd Karlsbakk wrote:
HELP NEEDED TURNING OFF THE cAPS lOCK KEY
:)
On Feb 25, 2005, at 20:07, Edward Banfa wrote:
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog
. Tomlinson
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
I'll look out for it, thanks!
Julius.
Julius,
I have just setup and installed phpconfig with the help of others on this
mailing list. I didn't use CVS checkout as I don't have CVS installed.
I
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius
Kidubuka
Sent: 28 February 2005 04:50
To: C. Tomlinson
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
I'll look out for it, thanks!
Julius.
Julius,
I have just setup
Its now up at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
I would be interested in any feedback. Hope it helps.
I've checked with IE and the numbers on the right point to extensions
defined in the file you are editing. That's a pretty nice feature.
The problem I
Wow,
The mediatrix configuration tru the unit manager software is like going
tru hell.My mediatrix unit is failing when it come to registering it
self with the asterisk box. I have set the realm to asterisk but still
nothing.
Does any one know what proper MIB variables to set and their full path
The mediatrix configuration tru the unit manager software is like going
tru hell.My mediatrix unit is failing when it come to registering it
self with the asterisk box. I have set the realm to asterisk but still
nothing.
Does any one know what proper MIB variables to set and their full path
Hi, thanks for the reply,
I finally got asterisk to register my mediatrix, I am now able to dial
an analog phone (connected to the mediatrix, which in turn is connected
to *) from a soft phone (X-lite).I made a typo when creating the
corresponding user on my asterisk box, after i corrected that,
HELP NEEDED TURNING OFF THE cAPS lOCK KEY
:)
On Feb 25, 2005, at 20:07, Edward Banfa wrote:
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius
Kidubuka
Sent: 25 February 2005 14:33
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
I am having trouble using cvs, is it possible to use cvsup
Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit.
This was true with the earlier SIP and H323 software, however if you get
their latest (11.70) it seems to be one of the better IADS out there. I do
agree with you though
Does this mean I have to download and re-compile my asterisk sources
inorder to get that file? And if yes, how do I get the sources with cvs
checkout phphconfig? If no, how is it done?
No, only do the cvs checkout phpconfig, and put the files in the right
directory that's all.
Guido Hecken
I am having trouble using cvs, is it possible to use cvsup or any other
method available and still get to install, configure and use phpconfig? If
so, how do I go about it?
Julius.
Does this mean I have to download and re-compile my asterisk sources
inorder to get that file? And if yes, how
PROTECTED] On Behalf Of Julius
Kidubuka
Sent: 25 February 2005 14:33
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
I am having trouble using cvs, is it possible to use cvsup or any other
method available and still get to install
Hi
On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote:
Secondly, is the statement no.2 a line a need to change in a given file?
You have to change/verify some settings in phpconfig_init.php .
Look for fakeuser=admin.
Set $reset_cmd = ./asterisk.reload;
Be shure, the script has
-
Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Gesendet: Freitag, 25. Februar 2005 16:25
An: asterisk-users@lists.digium.com
Betreff: Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Hi
On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote:
Secondly, is the statement no.2 a line
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog phones that i would like to connect
to my Mediatrix box and then connect the Mediatrix box to my asterisk
box. My main problems come from the fact that I have limited experience
Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit. First of
all - do you have the Mediatrix Unit Manager software? If not,
configuration will be nearly impossible. Secondly, you will need to
configure the sip ports on the
2005 06:59
An: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Betreff: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Thanks for your contribution Guido!
Do you have a URL I can visit to help me install and configure
phpconfig.php? Otherwise, I'll take a look at phpmyedit
: asterisk-users@lists.digium.com
Betreff: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Thanks for your contribution Guido!
Do you have a URL I can visit to help me install and configure
phpconfig.php? Otherwise, I'll take a look at phpmyedit and see how
ot
works for me.
Rgds
: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Do I have to cd into my asterisk source directory (that is,
/usr/local/etc/asterisk) or otherwise?
Yes, cd in your asterisk source dir, from where you installed it.
Secondly, is the statement no.2 a line a need to change in a given file?
You have
I do follow all the instructions except for one; I don't seem to have the
phpconfig_init.php file in my asterisk source directory. Maybe I have
overlooked something or forgotten to install a certain package.
If you get the sources with cvs checkout phpconfig, you should have this
file.
I
I do follow all the instructions except for one; I don't seem to have
the
phpconfig_init.php file in my asterisk source directory. Maybe I have
overlooked something or forgotten to install a certain package.
If you get the sources with cvs checkout phpconfig, you should have this
file.
Try [EMAIL PROTECTED] it has a great GUI that auto
installs
http://asteriskathome.sourceforge.net
--- Julius Kidubuka [EMAIL PROTECTED] wrote:
I do follow all the instructions except for one;
I don't seem to have
the
phpconfig_init.php file in my asterisk source
directory. Maybe I
Thanks for the advice though I would really love to get to the bottom of
what am onto right now, that is, try to find a solution for my current
Asterisk setup.
I shall most certainly try out [EMAIL PROTECTED], you can bet on that!
Rgds,
Julius.
Try [EMAIL PROTECTED] it has a great GUI that
On Mon, Feb 21, 2005 at 08:19:58AM +0300, Julius Kidubuka wrote:
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing my GUI using;
: Donnerstag, 24. Februar 2005 01:54
An: asterisk-users@lists.digium.com
Betreff: Re: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
On Mon, Feb 21, 2005 at 08:19:58AM +0300, Julius Kidubuka wrote:
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please
visit
http
: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
On Mon, Feb 21, 2005 at 08:19:58AM +0300, Julius Kidubuka wrote:
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please
visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing
-users@lists.digium.com
Betreff: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI
Thanks for your contribution Guido!
Do you have a URL I can visit to help me install and configure
phpconfig.php? Otherwise, I'll take a look at phpmyedit and see how ot
works for me.
Rgds,
Julius
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing my GUI using;
http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl
I am trying to get the menu options in my GUI to
hi all
i have asterisk on a TE410P card running on 2 E1s. E1(a) and
E1(b)
Following is the call flows
1.user call from POTS to E1(a) 2. asterisk will authentic
user ANI and once authentic is clear it will play DTMF A tone3. user key in
destination number4. asterisk uses LCR to find which
In a nutshell:
used to have SQL for SIP phones and Voicemail.
Upgraded to latest asterisk.
I am unable to puzzle out the docs for what I need to put in extconfig,
and what I need to add to my system for this to work like it used to
prior
to the update.
For example, I have database friends,
Hi all,
I am an MTech student and currently working on a project on GSM air
interface. I am making use of Asterisk soft PBX. I am stuck at a point
regarding this. As far as I understood from the available Asterisk
documentation that Asterisk can easily plug into it the various programming
On Wed, 8 Sep 2004 18:58:11 +0530, Renu Rangnekar
[EMAIL PROTECTED] wrote:
As far as I understood from the available Asterisk
documentation that Asterisk can easily plug into it the various programming
interfaces and different codecs in it can seemlessly talk to one another.
Asterisk has a
Title: Message
My Requirements
are
1) I need to
simulate a commercial level PBX ( Handling about 250 simultaneous calls )
using Asterisk.
Is this possible?
If yes then what is the hardware I need to procure for such an
Installation.
Moreover would Asterisk be able to handle such a
Hello List,
I'm from Germany and I want to use a Asterisk
System.
I have a few Accounts at my SIP-Provider www.sipgate.de and now I want to use my
ISDN-Phone on the Sip-System.
My idea was i set up a Asterisk-System and i will
put in an ISDN Card where I can plug a ISDN Phone, I will have
/
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Beierlein Moritz
Sent: Wednesday, July 21, 2004 12:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Help needed for Seting Up Asterisk
Hello List,
I'm from Germany and I want to use a Asterisk System.
I have a few
)
From:Beierlein Moritz [mailto:[EMAIL PROTECTED]
Sent:Wed 7/21/2004 15:04
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Help needed for Seting Up Asterisk
Hello List,
I'm from Germany and I want to use a Asterisk System.
I
phone -- the cost of the ISDN card
just isn't worth it. Get yourself a Sipura, and you can even do
distinctive ring and such.
-Original Message-
From: box100 [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 21, 2004 8:53 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help needed
Il 15:54, venerdì 09 luglio 2004, Andrew Thompson ha scritto:
Shanmuganathan Kumaravel wrote:
If anyone knows it pls help it would be very helpful regarding my
project work.
Regards
Shan
I've got the same issue between ata-186 and grandstream, this was a codec
issue to diagnose
I bought a Cisco 7940, I need to configure it for
Asterisk. I checked the wiki pages. Followed the link
to Cisco web page. Tried to download the image for
SIP. It wo'nt allow me even though I registered for
the CCO Valet. Is the image available anywhere else?
I saw some of the messages in the
The configuration for the 7940 is the same as the Cisco 7960, look at
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx and
http://www.voip-info.org/wiki-Setup+SiP+on+9740+-+9760.
Quoted from the wiki:
Note: Cisco software images are only available from Cisco's web site
and are protected by
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