Hey Chris,
Starts from here,
https://wiki.asterisk.org/wiki/display/AST/Getting+Started or try Asterisk
Complete guide in pdf format. If you are looking for something graphical,
go for elastix or freepbx.
Thanks
~Arun
On Thu, Nov 24, 2016 at 12:28 AM, christopher kamutumwa <
Hello all,
Im using a GSM gateway device for making outbound calls. GSM device is
connected to one of my SIP peer. Now am facing a lot of voice signal
problems. I checked with my vendor and there is no issues with signal and
device. Any settings in asterisk?
Thanks
Arun
--
Hi,
Change the protocol from tcp to udp in iptables.
~Arun
On 27 Jun 2014 20:07, Anurag Rana anuragrana31...@gmail.com wrote:
Hi All.
Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop it because I
am unable to detect the IP address.
Hello All,
I have a Digium Wildcard TE410P Quad-Span T1 Card, when I do connect T1
lines it goes in RED. When I do connect the same line on a different Server
(Same Model T1 Card) it works fine. How do I examine/diagnose my T1 Card
for any hardware failures. I heard about loopback test , how
Thank you Josh for your valuable reply. I will do try changing the server
and let you know what happening.
~Arun
On Tue, Jun 24, 2014 at 8:39 PM, Josh Metzger joshdmetz...@gmail.com
wrote:
On Tue, Jun 24, 2014 at 5:25 AM, arun kumar arunvsadni...@gmail.com
wrote:
Hello All,
I have
Cables are workig fine in my other box.
On 25 Jun 2014 00:46, Steve Totaro stot...@totarotechnologies.com wrote:
Remember to always check your cables first.
Thanks,
Steve T
On Tue, Jun 24, 2014 at 1:47 PM, arun kumar arunvsadni...@gmail.com
wrote:
Thank you Josh for your valuable reply
Its an old box with Asterisk 1.2
On 25 Jun 2014 03:46, Mc GRATH Ricardo mcgra...@mail2web.com wrote:
Why you configure zaptel.conf? should configure on dahdi files
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
--
Hi Guys,
Soon, I'll be starting a new section related to Asterisk (around 4 years of
full time experience with Asterisk, Trixbox, SER, OpenSer, MediaProxy, AGI*)
so let me know if you like to see some topic coming.
Cheers
Arun
--
HI All,
I got solved this issue.
Thanks all for your help
Arun
On Sun, May 24, 2009 at 1:58 AM, David Backeberg dbackeb...@gmail.comwrote:
On Wed, May 20, 2009 at 12:44 AM, Arun Kumar arunv...@gmail.com wrote:
here is my problem: when I call from 6004 to my cme extension 4615, on
4615
Hi All,
please provide some help.
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones Asterisk SIP Trunk Call Manager - CME
-
Hi All,
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones Asterisk SIP Trunk Call Manager - CME
- 461X Phones
461X Phones
Hi guys
I am working in Kanpur, India.
When someone calls to my server i forward the call to someone else by Dial
command. After dialing it says Native bridging. And after that I am unable to
detect whether the call was answered, the called number was busy or the call
was not completed.
One
-- Forwarded message --
From: Arun Kumar Chaudhary [EMAIL PROTECTED]
Date: Sat, Jun 21, 2008 at 4:51 PM
Subject: Detection of Answer, hangup,busy etc while using Dial command
To: [EMAIL PROTECTED]
Hi Guys,
I am in kanpur, India.
I am using Dial() command in my phpagi script. I am
Hi,
I've two wifi-phones
1. Nokia e65
2. HP Ipaq
I've configure two sip exten in my asterisk and using these exten in my
phones. But my Nokia phone is keep on loosing the connectivity very soon
life 1-2 min the qualify packet will be double of my HP. So, when I try to
call my Nokia SIP exten
try to use http://www.fring.com/download/
On Nov 21, 2007 3:28 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Here's one sip softphone for mobiles you can give a try:
http://www.minisip.org/
Regards,
Ricardo Carvalho.
___
--Bandwidth and
Hi
here is my setup :
1. USER - PSTN - Asterisk A - IAX2 Trunk - Asterisk B - SER -
Asterisk C (Accepting DTMF)
All Asterisk box has dtmfmode = inband, when user pressed DTMF able to
receive and working fine.
2. Asterisk C --- Dial Customer
Customer input DTMF and its not taking any dtmf but
Hi
Here is my setup:
USER -- PSTN - Asterisk A IAX2 Trunk Asterisk
B - SER Asterisk C
I'm not able to receive DTMF passed by USER on Asterisk C.
All my asterisk boxs are configured with same DTMF type (auto) but no luck.
Please help on this issue.
just configure SER on another port and use.
On 11/1/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
anybody have implement SER with Asterisk in single machine ?? i
have asterisk with 200 SIP device but i voice qulity and load of asterisk is
bit high so i need to implement SER
try to reduce number of calls on trunk or create multiple trunks.
On 10/31/07, Louis-David Mitterrand [EMAIL PROTECTED]
wrote:
Hi,
Using 1.4.13 and trunking a single iax channel to a similar box my
asterisk console is flooded with:
[Oct 31 10:49:34] WARNING[5195] chan_iax2.c:
Hi
Need help on this setup:
Incoming DID in H323 Asterisk Server -- SIP Phone
please tell me to achieve this above setup what needs to be done in
Asterisk.
thanks
Arun
___
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Hi
I've configured my asterisk and voicemail all works fine but I want to
restrict call time to voicemail that is when user calls voicemail he
can use voicemail system only for a max of 5 min that is after five
minutes asterisk should disconnect the call.
thanks
Arun
Hi All,
UA Asterisk Server - UB
if there is no rtp for a specified number of minutes / seconds then I want
to disconnect the call. I've tried using rtptimout and rtpholdtimeout but no
luck
pls guide.
thanks
arun
___
Sign up now
run iax2 show peers and see next to port (T) if it comes then you are using
IAX2 Trunking feature.
On 8/10/07, George Pajari [EMAIL PROTECTED] wrote:
How can one verify that IAX trunking is in effect and that Asterisk is
trunking multiple call paths between two Asterisk servers?
With 1.4.10
Hi
I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI
scripts are not working properly. Like after hangup I used to do some more
work now its not working.
Please help.
thanks
arun
___
--Bandwidth and Colocation Provided by
HI
Here is my info:
Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents
this asterisk box is connected to another asterisk box using 5 IAX trunk to
load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my
cli start flooding with message: Maximum trunk data space
issue is got solved by moving to another pri card and now congestion works
fine with my ISP.
thanks all.
On 7/18/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On 7/17/07, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
I did a quick test.
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
Hi
using php script and Asterisk manager I'm dialing numbers and once gets
connected send to an exten in my dial plan that plays an automated message
but some time without answering even it goes to my exten. How can I handle
early media in Asterisk that is I want only when user answer the call
Hi
I need help in configuring a auto dialer system using Asterisk. I'm holding
my customers number in MySQL want to fetch 10 numbers one time and dial if
gets connected and answered by customer wants to play a sequence of message
. Please help .
I've tried here is my code to place calls but
Hi
this is my code for * manager:
$oSocket = fsockopen($strHost, 5038,
$errnum, $errdesc) or die(Connection to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Username: $strUser\r\n);
thanks for reply. I've same setup with siml. incoming calls 10-12 it works
fine but some time my machies get hang and gives same IAX max data space
error.
thanks
On 6/27/07, Jared Smith [EMAIL PROTECTED] wrote:
On 6/27/07, Arun Kumar [EMAIL PROTECTED] wrote:
so , how much bandwidth I need
Hi
I've two servers :
1. UK
2. Pakistan
Pakistan * server has ISDN30.
Pakistan(ISDN30) UK === User
Im planning to setup an IAX2 trunk between these two server ?
so , how much bandwidth I need for 30 simul. calls ?
Im planning to use G729 on both my server ?
to support 30 calls over
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to
use IAX2 trunk.
On 6/7/07, Sebastian Reitenbach [EMAIL PROTECTED] wrote:
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing
Hi
I've two boxes connected over IAX2 trunk before IAX I was using SIP trunk
and they were working fine b'coz of bandwidth issue I changed from SIP to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I see lots of messages of IAX2 trunk and b'coz of that my
On 6/5/07, Noah Miller [EMAIL PROTECTED] wrote:
Hi Arun -
I've two boxes connected over IAX2 trunk before IAX I was using SIP
trunk
and they were working fine b'coz of bandwidth issue I changed from SIP
to
IAX now I'm facing a strange problem after some time on the cli of my
asterisk box I
HI
I'm looking for a card that support both PRI and TDM. Please suggest me ?
thanks
arun
___
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To UNSUBSCRIBE or update options visit:
Hi
I've two boxes connected over IAX2 trunk but suddenly my cli is getting
flood with these messages:
iax2_trunk_queue: Maximum data space exceeded
and b'coz of that my agents are not able to hear any thing.
when this happened that time there were 9 calls.
my * version is 1.2.18 and 1.2.14
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in my old box also ?
thanks
arun
Hi
I've two boxes connected via IAX2 Trunk were working fine from few days
suddenly today one box is got crashed with this message
2007-06-03 12:25:37 WARNING[26511]: chan_sip.c:2612 sip_write: Can't send
4113608 type frames with SIP write
my version of * is 1.2.14 on FC4
thanks
arun
HI
Im getting strange message on asterisk console
WARNING[26853]: app_queue.c:2321 try_calling: Announcement file
'custom/announce-adslsetupnatrate' is unavailable, continuing anyway...
thanks
arun
___
--Bandwidth and Colocation provided by
Hi
my * box is giving me these warning and b'coz of second warning line my
agents are not able to hear the announcement in the queue some time it
happen many time
2007-06-03 13:40:30 WARNING[28016]: chan_sip.c:2612 sip_write: Can't send
4113568 type frames with SIP write
2007-06-03 13:40:30
Hi
sorry for not details. when ever I see this message on * console my agents
are not able to listen to announcement.
thanks
arun
On 6/3/07, Mattt [EMAIL PROTECTED] wrote:
And you don't find that sufficiently self-explanatory?
On Sun, 2007-06-03 at 13:02 +0400, Arun Kumar wrote:
HI
Im
-- Forwarded message --
From: Arun Kumar [EMAIL PROTECTED]
Date: May 13, 2007 5:40 PM
Subject: TC400B load problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi
Im trying to install my TC400B trans coder card when I do
thanks Matthew, I'll try to call Digium.
On 5/14/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
On May 14, 2007, at 4:53 AM, Arun Kumar wrote:
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=000c, dsts=0101)
May 13 14:56:36 pbx2 kernel: Registered codec translator
hi
vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c
this file and look for line that says 2.6.19 change it to 2.6.18 and save
and compile
arun
On 5/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, May 04, 2007 at 01:55:20PM -0400, mail-lists wrote:
I get the following error when
Hi
I already tried asterisk manager but Im not able to get status for each
queue member.
thanks
On 5/8/07, Edoardo Serra [EMAIL PROTECTED] wrote:
Hi,
you can use an AGI to connect to asterisk manager and retrieve the
info you need about the queue.
Hope it helps
Arun Kumar ha scritto
Hi
I've few queues configured in * box is there any what that before sending
call to a particular queue can we get the status of the queue that is how
many agents are available in this queue (logged in, paused, busy,
unavailable).
thanks
arun
___
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Arun Kumar
*Sent:* Tuesday, May 01, 2007 9:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Change Codec
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
Hi
this is my setup:
Customer - PRI - Server A with G729 - IAX2 Trunk(G729) - Server B
- SIP Exten allowed codec=g729 - Snom phone Agents
setup is working fine.
I want when my agents are not available (queue) like not logged in or all
are busy so no calls should come to my server b from
Hi
I've few queues configured in * box is there any what that before sending
call to a particular queue can we get the status of the queue that is how
many agents are available in this queue (logged in, paused, busy,
unavailable).
thanks
arun
___
Hi,
Is there any way that I can store my manager API output that is:
My question is that is there any why using that I can get the QueueStatus
and store the result in some text file for further processing.
?php
$strHost = 127.0.0.1;
$strUser =
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Arun Kumar
*Sent:* Tuesday, May 01, 2007 9:24 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Change Codec
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
Hi
I've install Asterisk 1.4.2 and its working fine. In my sip.conf I've
allowed ulaw and g729. I want to change the codec for outbond calls. Please
help not able to find anything using search.
thanks
arun
___
--Bandwidth and Colocation provided by
In my * box I've configured two queues and incoming number and whenever any
one calls those number call comes to my *box and it sends call to my agents
in queue. but if no agent is available it still answer the call. Is there
any why when my agents are not available I don't want call to get
to help further.
On 4/24/07, Arun Kumar [EMAIL PROTECTED] wrote:
Hi,
I'm running a php script to generate calls using Asterisk Manager and
its working fine. this script call a specified land line number if the phone
is answered then It will connect to an extension and play an IVR. But I see
Hi,
I'm running a php script to generate calls using Asterisk Manager and its
working fine. this script call a specified land line number if the phone is
answered then It will connect to an extension and play an IVR. But I see in
Asterisk CLI its placing the call and it shows channel answered
Hi,
I've configured my exten.conf for few exten. But I'm curious to know how
long can be my exten like (exten = XXX.). Is there any limit for
this or not. B'coz I've noticed one strange problem. I'm usnig snom300 as my
hard phone to make calls. when my exten length is 14 then calls goes
Hi,
in my dial plan I've configured two trunks to make outbound calls (one for
national calls and other international). I want to allow only 2-3 extension
to make use of my international trunk to make outbound calls so I want some
kind of auth. based on their callerid . Please guide.
thanks
Hi
I've configured the queue on my asterisk box and everything is working fine.
In my queue I've 3 agents logged in the queue. When call comes they are able
to receive the calls without any problem. But some time they are on break
and there extension rings and no one is there to answer the call
Hi,
Is it possible to design an IVR using SER ? If yes please advice.
thanks
arun
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To UNSUBSCRIBE or update options visit:
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: Arun Kumar [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Thomas Kenyon
[EMAIL PROTECTED]
Sent
Hi
sorry for asking the same question again:
here is my details:
I've 50 exten in my sip and I'm using snom300 to my asterisk box this
asterisk box is connected to another asterisk box using IAX trunk over 1MB
full duplex line. I'm using g729 as the preffered codec. Can you please tell
me how
://site.asteriskguide.com/bandcalc/bandcalc.php
On Tue, 17 Apr 2007 11:54:28 +0400, Arun Kumar [EMAIL PROTECTED]
wrote:
Hi
sorry for asking the same question again:
here is my details:
I've 50 exten in my sip and I'm using snom300 to my asterisk box this
asterisk box is connected to another asterisk box
how do I check that whether trunking is working or not ? No I don't any
timing soure (like zaptel card).
thanks
On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Arun Kumar wrote:
I've tried this but stil some problem Like if I use this link that you
gave me it shows for 10 call 136.08KBps
how do I check that whether trunking is working or not ? No I don't any
timing soure (like zaptel card) b'coz these are test server. what else I can
use for timing.
thanks
On 4/17/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Arun Kumar wrote:
I've tried this but stil some problem Like if I use
Hi,
In my dial plan I've configured two trunks to make outbound calls (trunk1
and trunk2) to same service provider but I want when any of my exten starts
with _2. should goto trunk2 and there should be some kind of disturbance
(like some noise or some background noise) when my calls goes to
HI,
Here is my setup:
USERS - PSTN - Service Provider - Asteriskbox1 - IAX2 trunk - Internet
- IAX2 trunk - Asteriskbox2 -Sip Clients
between asteriskbox1 and asterisk box2, I've VPN configured. from
Asteriskbox2 to internet my line speed is 1MB.
Is there any why that I can calculate how
Any help appreciated
Thanks!
Rajeev
On 2/20/07, Arun Kumar [EMAIL PROTECTED] wrote:
Instead of forwarding to IAX softphone if I'll play some music same
thing is happening in this case also.
On 2/20/07, Mark Phillips [EMAIL PROTECTED] wrote
HI
I've configred an Incoming DID in my asterisk and when I call from outside I
see call is coming to my Asterisk server and then from asterisk it rings on
a particulat exten but when I pickup the call the call get disconnect
immediate and on the other end it keep trying (ringing).
here is my
phone is using ulaw and your DID provider is using
something else like G729.
Mark
On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote:
HI
I've configred an Incoming DID in my asterisk and when I call from
outside I see call is coming to my Asterisk server and then from
asterisk it rings
use LCR is really good.
On 1/22/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Users,
How can I perform the load Balancing in My SIP server of Both OpenSER
and Asterisk ,
Currently I have One OpenSER server and Asterisk Server,
For OpenSER is to need use these modules, and is
Hi All,
Wish you a very HAPPY and Merry Christmas to all and your beloved once.
Arun
On 12/23/06, Josué Conti [EMAIL PROTECTED] wrote:
Hi ALL,
** I like very to desire you and your family, a Merry Christmas, with much
love, peace, professional and personal success.
Best Regards
Josue
HI,
My Asterisk is registed with my SER. My client are connected to asterisk
when they dial any no like 6 asterisk passes this is ser and then again
ser passes this no (strip 1) back to my asterisk. but insted of ringing
this exten it says loop detected. can some one tell me what is
HI,
thanks for your reply. Here is my ser.cfg and other config files please
guide me.
ser.cfg
--
debug=5
fork=no
log_stderror=yes
listen=2xx.xxx.xxx.xxx # INSERT YOUR IP ADDRESS HERE
port=5060
children=4
dns=no
rev_dns=no
fifo=/tmp/ser_fifo
fifo_db_url=mysql://ser:[EMAIL
HI,
I'm not able to find some good doc or manual regarding Integration of
Asterisk with SER. Bacially, I want to forward my calls from SER to
asterisk. If some one already done this please guide me.
thanks in advance
arun
___
--Bandwidth and
hican you please post some user or config guide.thanks in advancearunOn 10/24/06, Ed Greenberg
[EMAIL PROTECTED] wrote:I will sign in with good experiences with MP124 and Mediant 1000. I have an
MP202 under test.--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas[EMAIL PROTECTED] wrote: I
can please some one tell me where is what wrong.
iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/callaus-3
can please some one tell me where is what wrong.iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO KpktsIAX2/callaus-3 265 -1 0 -1 -1 0 -1 00 40 0 0
b'coz I have same setup at other client is working fine no problem.On 9/22/06, Tzafrir Cohen [EMAIL PROTECTED]
wrote:On Fri, Sep 22, 2006 at 03:09:47PM +0530, Arun Kumar wrote: can please some one tell me where is what wrong.
iax2 show netstats LOCAL
HiI've * running but I'm other side voice is not so clear and delay. this is my iax netstat output can someone help me where is the problem.here is the iax netstat output Channel RTT Jit Del Lost Drop OOO Kpkts Jit Del Lost Drop OOO Kpkts
3 Traffic from Server to Agent 4 IAX2/2003-29 6 -1 0 -1 0
no zap - iax2 - iax2 only iax2 - iax2 - iax2thanksOn 9/21/06, Ma Zhiyong [EMAIL PROTECTED]
wrote:I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK.
___--Bandwidth and Colocation
hithanks for reply.I'm using vicidial to make calls at 2.0 dial level it is able to make calls but when I see the asterisk -r most of the time it shows Outgoing Spool Failed. Which Spool File ?thanks
arunOn 9/8/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-Hash:
himy asterisk -r shows me Most of the times Outgoing Spool Failed. Can some one tell me why is it happening and how to solve this issue. Is it a problem ? thanks in advance.arun
___
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asterisk-users
hi,can you describe what you want.../ArunOn 7/14/06, varun [EMAIL PROTECTED]
wrote:Hello,We were able to get asterisk going withX100p cards on centos
4.2.But could on centos 4.3 due to kernelissues.Anybody has faced this issue ?And how do sort it out so that wecan use centos 4.3
Hi,Here are the step by step instructions for setting up a brand new AudiocodesFXS gateway for use with an Asterisk server: Connect the gateway to a network switch and connect a computer to the same
switch. Then configure the IP address of the computer to 10.1.10.2. Then runyour web browser and
Hi,When you originate a call asterisk essentially callouts to the
Specified channel and the when answers connects the the
context,extension,priority. What if I want my dial plan to make the
origination call and the destination call. What would I specify for my
dialplan/callout file?thanks in
Hi,you check your a2billing.conf file:; Please enter here the file you want to play when we prompt the calling party to enter his destination number; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011
file_conf_enter_destination = prepaid-enter-destI think this file should help
Hi,
What is the Location. I'm studying in India. Is it possible.
thanks
ArunOn 6/17/06, Bob Knight [EMAIL PROTECTED] wrote:
I have 4 sparc based sun boxes I am about to pay money so I canget rid of them.They are running older versions of Solaris.You should be able to load Solaris 10 and play
Hi Calvis,
Its good if I can help you in any why with this project.
thanks
../ArunOn 6/2/06, calvis [EMAIL PROTECTED] wrote:
Has anyone done any integration with Asterisk Microsoft Dynamics CRM?Ijust wanted to check with the list before I pursue a project with the aboveintegration.In addition,
will need PhoneJACK (PCI or ISA)
Few days ago they announced that there is a new PhoneJACK PCI available
- with new DSP and etc.
Best regards
Lubo
Arun Kumar Sharma, Noida wrote:
Thanks Adam,
This document provides me a high level architecture of Asterisk. Can you
please tell me if I want
Hi Everybody,
I am new to Asterisk. Can anybody suggest me some link where I can find
architecture level detail of this system. My aim is to find out how easy it
is to port it on a new hardware (T1/E1 and POTS)?
Any input is highly appreciated.
Regards
Arun
-Original Message-
From:
Message-
From: Low, Adam [mailto:[EMAIL PROTECTED]
Sent: 17 July 2003 19:11
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] [Asterisk-Users]Help Needed
http://www.digium.com/handbook-draft.pdf
-Original Message-
From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED]
Sent
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