+++ Peter Bowyer [17/04/06 06:57 +0100]:
On 17/04/06, Vikram Rangnekar [EMAIL PROTECTED] wrote:
you can fix issue number 3 by running the install script
sh ./install.sh
or manually running the command
touch /var/log/asterisk/druid
chmod 777 /var/log/asterisk/druid
You'll have
+++ Doug Lytle [16/04/06 18:23 -0400]:
Vikram Rangnekar wrote:
+++ Doug Lytle [11/04/06 07:58 -0400]:
Vi
I hope you can reconsider your decision now.
I have,
I setup a test system with a copy of my configurations. Here are some
suggestions:
1.) Put together
Hi All,
Voiceroute http://www.voiceroute.net has released a new version of the DRUID
Web-interface for Asterisk. There is a free trial edition available as well
as a live online demo both can be accessed through the website.
Tons of new features and extensive use of Ajax makes it a very easy
and
+++ Doug Lytle [11/04/06 07:58 -0400]:
Vikram Rangnekar wrote:
Feel free to try it out and send us any feedback you may have.
Vikram,
A few issues.
1). Requires to be run on the Asterisk server via Apache. On a
production machine, I try to keep the services to a minimal.
2
I was checking out the asterisk-netsec code today and was suprised that it
dosent even comes with a sample midcom.conf file.
Also documentation is quite poor. If a company wants to release its code to
help Asterisk it should maybe also add a little documentation.
Just a file explaining what each
+++ Strain Jer [30/01/06 01:29 +]:
I was searching thru the internet and I found a wide variety of different
web interfaces for asterisks
I was curious which one is best suited for asterisks. Thanks
___
--Bandwidth and Colocation
Hi,
We have recently setup a Live Demo of DRUID our Asterisk management interface
product. Also I'd like to thank all of you that took the time to download the
trial edition and give us your feedback. WE've tried to incorporate as much
of that feedback into our new updated release.
Feel free to
Hi Guys,
We got a lot of feedback asking for free trail editions so here it is. A full
featured trial. We are definatly commited to this product and are focused on
creating the best web/non-web interfaces into asterisk for easy management of
remote and local Asterisk servers.
Your support and
Hi,
Druid is a new Web-based Asterisk management software. Its quite feature
packed and allows you to manage every aspect of Asterisk configuration. It
also has a Java Applet based Manager Console so you can visually monitor what
your Asterisk box is upto.
We will have a live demo up soon but
Hi guys,
Does anyone know of a way where I can bring a third person in on my
conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM
and am speaking to someone now I want to quickly bring another SIP or IAX
extension into this call so the three of us can speak to each other.
I am currently installing an Audiocodes MP-108 with Asterisk in Singapore.
I am able to make outgoing calls but incomming calls just get cut by the
MP-108
I can see the corrosponding frontpanel led light p but the call gets cut so
I'm very confused about what could be wrong.
In a pure voip envoirnment which uses a single codec say ulaw across all its
phones can asterisk support 5000 voip sip phones on a dual / single xeon with
1 gb ram. If all the phones support reinvite (Send RTP stream directly to
each other).
Or would I need more than 1 system to support 5000
Hi,
Ya I know about SER but I wanted to specifically know about asterisk in this
envoirment and its always great to stress test asterisk any bugs that comeup
would help make asterisk better.
+++ [EMAIL PROTECTED] [27/05/05 12:13 +0200]:
Hi,
Asterisk will work, but in your situation I think
+++ Mark Johnson [20/05/05 11:08 -0400]:
Ok, guys... Please be gentle with me. I have what is going to be the
strangest question you will have ever heard, but I have no idea what to
tell this person.
I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My
receptionist has
Sangoma is great prompt response to the problem, its fixed now
Use the latest stable latest stable 2.3.2-3
+++ Vikram Rangnekar [05/05/05 08:58 +0200]:
have an asterisk box (P4, 1GB RAM, etc ) with a sangoma two E1 port card in
it. I have used a E1 cross cable to connect the two E1 ports
I have 20+ asterisk servers and need to network them together so a phone on
any of the servers can call a phone on any other server without any trouble.
I can think of IAX trunks between every server. So every server will have an
IAX trunk to every server and then prefix bases routing in the
+++ Alex Barnes [09/05/05 15:26 +0100]:
But this doesn't sound particularly elegant specially once you start
trying to scale it.
If you do get any other ideas I would be interested to know so that I
can start this
structure out properly.
Your right using dialplan extensions for every
+++ Kanuri, Seshu (Company IT) [09/05/05 11:25 -0400]:
Vikram,
Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.
There may lie a plausible solution for you. If this is done, you may go
and
+++ Sam Njenga [09/05/05 13:55 +0300]:
why not try using extensions from a database on all servers. That way
regardkess of the * server, the destination phone is the same.
http://www.voip-info.org/wiki-Asterisk+RealTime
I think you might be on track but again I dont know how stable
+++ Vikram Rangnekar [09/05/05 12:25 +0200]:
I have 20+ asterisk servers and need to network them together so a phone on
any of the servers can call a phone on any other server without any trouble.
I can think of IAX trunks between every server. So every server will have an
IAX trunk
have an asterisk box (P4, 1GB RAM, etc ) with a sangoma two E1 port card in
it. I have used a E1 cross cable to connect the two E1 ports together and
pass calls through them for testing purposes allo works fine. also have
multiple voip phones connected to this setup,
what i noticed is that
+++ Peter Svensson [05/05/05 12:43 +0200]:
On Thu, 5 May 2005, Vikram Rangnekar wrote:
what i noticed is that when i pull any one end of the E1 (breaking the E1
connection) I get multiple RED ALARMS on the zap channels I understand this
is ok and should happen if the E1 link breaks but my
EPBX's in
India which really act wierd sometimes.
On 4/15/05, Vikram Rangnekar [EMAIL PROTECTED] wrote:
Just so that you know that would be considered illigal in India if you are
planning to have VIP extensions on that Asterisk install also. Many people
have been raided and even sent
+++ Min Hwan Chang [15/04/05 01:29 -0700]:
I'm currently trying to set up an Asterisk PBX system in India.
However I'm having trouble configuring the X100P to dial out on the
POTS line. Does anyone have any knowledge about this?
I know the telephone system is a bit different in India, so
+++ Min Hwan Chang [15/04/05 11:25 -0700]:
My Setup is below:
X100P connected to POTS
Siemens IP Phone
[EMAIL PROTECTED] v0.6
When I try dialing out the POTS line in India using a Siemens IP
phone, I'll hear the phone ring for a second, then hear loud beeps.
The loud beeps I believe
Does anyone know the exact VOIP regulations in India. What I want to know is
that are VOIP EPBX with E1 lines allowed for comapnies in India. For example
If I am a company which has 1 incomming E1 line can I have SIP or IAX
extensions inside my office and receive that PSTN call on the VOIP
http://www.trai.gov.in/IP_Recommendations.htm
Section 3 and 4 of this article talk about how the digital divide will
increase if ISP's are allowed to provide PSTN VOIP integration by cutting
into the revenue stream of other Telecom providers who are obligated to
provide low cost telecom services
+++ Dan Austin [13/04/05 14:45 -0700]:
http://www.trai.gov.in/IP_Recommendations.htm
Section 3 and 4 of this article talk about how the digital divide will
increase if ISP's are allowed to provide PSTN VOIP integration by
cutting
into the revenue stream of other Telecom providers who
+++ James Coberly [25/03/05 09:22 -0700]:
Vikram Rangnekar wrote:
+++ Dan [20/03/05 09:17 +0200]:
Hi James,
- Original Message -
From: James Coberly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent
+++ Dan [20/03/05 09:17 +0200]:
Hi James,
- Original Message -
From: James Coberly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 19, 2005 11:41 PM
Subject: Re: [Asterisk-Users] Re: Optional URL
+++ James Coberly [18/03/05 10:28 -0700]:
Try DIAX.
http://www.laser.com/dante/
Didnt work for me I logged into asterisk as an agent from diax but it didnt
open up a browser for the url i set in the queue options :(
Vikram Rangnekar wrote:
I have googled for days abt this so finally i
I have googled for days abt this so finally i turn to the list that knows all
:)
I want to use the Optional URL parameter in the Queue application to redirect
agents to a webpage. Now I cant seem to find a IAX client for windows that
supports it. I dont mind a SIP client either but I dont want to
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent: Tuesday, February 15, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED
Its fixed and working great.
I was working on the dma
information.
+++ Vikram Rangnekar [13/02/05 13:01 +0100]:
Does anyone have any experience ith configureing the sangoma A102 card for
testing using a e1 cross cable i've configured and installed the cards
properly even the lights on the card are green which proves that my cross
cable is properly built
Does anyone have any input into integrating asterisk with a alcatel 4400 PBX.
Acording to what i've found is that Alcatel uses R2 for E1
--
regards
Vikram (http://www.vicramresearch.com)
___
Asterisk-Users mailing list
Does anyone have any experience ith configureing the sangoma A102 card for
testing using a e1 cross cable i've configured and installed the cards
properly even the lights on the card are green which proves that my cross
cable is properly built too. my problem is with asterisk which gives me these
+++ Duane [13/02/05 22:56 +1100]:
On Sun, February 13, 2005 23:01, Vikram Rangnekar said:
span=1,0,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
span=2,1,0,ccs,hdb3
bchan=32-46
dchan=47
bchan=48-62
At a guess 16,32,48 and 64 are d channels, where as you are telling
to set
to make the card use an internal clock.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent: zondag 13 februari 2005 13:02
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sangoma A102 cards testing
Does
thanks you for the information its extremely useful.
+++ Peter Svensson [04/02/05 08:59 +0100]:
On Fri, 4 Feb 2005, Michael Bielicki wrote:
It's even easier. If you are taking an E1 with euroisdn, the D-channel
will allways be nr. 16. In neary all cases the rest of the settings
will be:
I am planning to go in for a E1 line and whould like to know what questions i
need to ask my service provider so i can connect that E1 to my asterisk box
using the digium E1 card.
what I mean is will my service provider give me info like LBO, framing ,
coding etc which i need to configure the
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is there anything being done to fix
it. Just wanted to know why SER claims to be better than asterisk as a
After 20 cups of strong coffee and wasteing most of tonight and obviously
doing lots of googling and emailing many people, i've concluded that dlink
voip phones specifically DPH-80 dosent work with asterisk.
If anyone has had anymore luck with these phones then me please let me know
--
regards
I am looking for some in India to buy VOIP phones from. Please get in touch
with me off the list on [EMAIL PROTECTED]
Sorry for the off topic mail I am just having such a hard time finding any
voip phones in India that I got desperate and didnt know which list to post
this on.
--
regards
+++ Jay Austad [13/12/04 03:49 -0600]:
Anyone have an easy fix for making my music on hold to work properly?
It's very loud and has a lot of garbling in it. X is not running, and
the framebuffer is disabled.
I've tried just about every example I could find. I just uploaded
standard
I just came across this phone MTA 3308 By Innomedia its documentation says it
supports MGCP 1.0 and SIP 2.0 anyone have any experience with this phone. I
specifically wanted to find out if it works with asterisk before i purchase
it. I checked the wiki and the lists dosent seem like anyone has
+++ Victor Cartes [04/11/04 13:14 -0400]:
Does anybody know how to stop the AGI process after the user Hang-Up?
'Cause it stills running if the transaction is ended by the user.
Thanks
V?ctor
___
Asterisk-Users mailing list
[EMAIL
I was wondering how the reload option in asterisk (asterisk -r -x reload)
affects calls in session and other activity like active AGI aplications. I
tried it using a single call which i placed to my asterisk box and it didnt
get disconnected when i reloaded asterisk.
But what about heavy load
+++ Benjamin on Asterisk Mailing Lists [31/10/04 18:11 +0900]:
On Sun, 31 Oct 2004 08:18:22 +0100, Vikram Rangnekar
[EMAIL PROTECTED] wrote:
Also does
frequent reloads affect the stability of asterisk i mean things does it lead
to things like memory leaks
Depends on the version
Does anyone have any experience with using ast_data and keeping the diallan
in a mysql database. I setup ast_data properly and also setu mysql with the
required tables. I've inserted two dialplan entires in the database.
I ran asterisk CLI with data debug on and can see ast_data querying the
+++ Gunnar Schaller [14/10/04 22:40 +0200]:
I have also ast_data and extensions in mysql, works fine for me. Your
crash might be something other, I dont't think it's ast_data. Do you
have more infos to the crash? Logfiles?
Gunnar
___
when I'm using asterisk with ast_data for storeing the extensions in mysql.
Asterisk segfaults and crashes when i call an extension thats listed in a
mysql database instead of the extensions.conf file.
I couldnt see these errors when i ran asterisk with asterisk -r only saw then
when i ran
+++ Tenorio, Leandro [15/09/04 10:58 -0300]:
I've seen a lot of times, people that try to get R2 MFC to *, most of them trying to
use Dialogic Boards (BTW They 're Very expensive), none of them where succesfully,
If you want to use PCI Cards on your server, why don?t u ask to your carrier to
+++ Victor Rini [06/09/04 18:12 -0700]:
Hello,
After poking and prodding at Asterisk and Zaptel for over a couple years
now, I've dedicated some time to actually reading the code and trying to
figure it out.
It's been fascinating. With the driver source on one part of the screen
and a
After days of searching i've finally figured out that E1 lines in india use
multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work
for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards
which would work on R2. Also if anyone on this list is from INDIA and
+++ Marcelo Pacheco [27/08/04 11:06 -0300]:
It's not the card's fault, it's the lack of a software driver fault.
R2 has a country dependent implementation. Some countries even have two
incompatible standards internally.
Em Sex 27 Ago 2004 11:03, Vikram Rangnekar escreveu:
After days
I just came across libr2 anyone using it in its current state. Specifically
someone from India or around India using it. Also does it work with the
digium e1 cards or only the Dialogic cards.
http://digium-cvs.netmonks.ca/viewcvs.cgi/libr2/
--
regards
Vikram (http://www.vicramresearch.com)
+++ Vikram Rangnekar [13/08/04 13:35 +0200]:
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com)
___
Asterisk-Users mailing list
Has anyone had any luck with voice recognisation using sphinx, if yes then
could u please send some pointers.
does the eagi app for sphinx really work cause i'v tierd it and sphix dosent
seem to do anything
--
regards
Vikram (http://www.vicramresearch.com)
I configured agi-test.agi on extension 111 when i dial into asterisk
extension 111 using a IAX softphone and hangup while the AGI is playing
asterisk crashes. Does anyone have any idea why this happens.
--
regards
Vikram (http://www.vicramresearch.com)
I need to setup asterisk so that users can dial into asterisk using normal
phone lines and and enter a number when prompted then this number should be
accessable to a backend app. is this possible in asterisk. any pointer would
be helpfule
--
regards
Vikram (http://www.vicramresearch.com)
+++ Ernest W. Lessenger [16/03/04 10:03 -0800]:
At 09:52 AM 3/16/2004, you wrote:
I need to setup asterisk so that users can dial into asterisk using normal
phone lines and and enter a number when prompted then this number should be
accessable to a backend app. is this possible in asterisk.
exten = 666,1,Answer
exten = 666,2,AGI(agi-text.agi)
exten = 666,103,Hangup
iwhy is that not working any idea. Does answer need to be there or does the
AGI script answer the call.
--
regards
Vikram (http://www.vicramresearch.com)
___
Asterisk-Users
For some reason voicemails are not being played back. I can log into the
voicemail system and i get the menu. its all fine till the point asterisk is
announcing info abt the mail but when it comes to playing the mail i hear
nothing then it quickly gones on to the next mail. If anyone has
When using the softphone DIAX I cant receive calls when the use IAX2 checkbox
is ticked. Has anyone else had this same problem.
--
regards
Vikram (http://www.vicramresearch.com)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
+++ Dan [18/02/04 23:51 +0200]:
Hi,
- Original Message -
From: Vikram Rangnekar [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, February 18, 2004 11:45 PM
Subject: [Asterisk-Users] IAX2 dosent work for DIAX
When using the softphone DIAX I cant receive calls when
[bombay]
include = features
include = internal-sip
include = internal-iax
include = bogus
[internal-sip]
exten = _2XXX,1,Dial(SIP/${EXTEN},20)
exten = _2XXX,2,Voicemail(u${EXTEN})
exten = _2XXX,102,Voicemail(b${EXTEN})
exten = _2XXX,103,Hangup
[internal-iax]
exten =
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