vi /etc/asterisk/extensions.conf
On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote:
hi,
how to write a advanced dial plan
for example:
dial to a extension(123).if the user didnot pick the call, caller should get
a ivr script(Enter 1 to to dial operator and 2 to go to
hi,
how to write a advanced dial plan
for example:
dial to a extension(123).if the user didnot pick the call, caller should get a
ivr script(Enter 1 to to dial operator and 2 to go to voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it should
go to the voicemail
Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf,
sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to
call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using
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Check your extensions.conf
On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote:
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Andrew Joakimsen wrote:
Check your extensions.conf
Hahahahaha!
Doug
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[EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We were personally thinking in
recommending either Cisco, Aastra, Polycom, or
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50 in
one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Wednesday, October 31, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote:
Hi all,
We have a client
:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be pretty
good. The Polycom work well, but they seem
Honestly, Its my opinion that the Aastra phones are very lacking in
the firmware department. If they could get that sorted out I wouldn't
mind using them. But for now there are too many NAT issues mostly
caused because they use an OLD version of Broadcom CallCtrl. Why they
use an ancient version
Discussion
Subject: Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be
pretty good. The Polycom work well, but they seem to die after about a
year or so. We bought 20 of them
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We were personally thinking in
recommending either Cisco, Aastra, Polycom, or Snom, for we've heard
great
Of [EMAIL PROTECTED]
Sent: Monday, October 29, 2007 10:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] (no subject)
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model
Stay away from Cisco they just don't work for the price, if it would
be in the price range of a Grandstream phone I would tell you go for
it, but at the current price its just not worth it. Aastra, Polycom or
linksys all work for me. Never tried Snom before.
On 10/29/07, [EMAIL PROTECTED] [EMAIL
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] (no subject)
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We were personally thinking in
recommending
Hello,
I am looking for an Asterisk consultant for occasional support on an
asterisk phone system located in San Francisco. It would probably be
primary remote support, but we may need some on site support
occasionally. Please let me know if you are interested and available.
Thanks,
Motherboard with SATA RAID1 support
That's a mulit-port SATA controller with RAID in the driver (software).
256 MB RAM
Use a little more RAM.
digium PRI/E1 card
Is there any reason you aren't using Sangoma cards?
1. If I use Software RAID, what would be the impact to my deployment? (
On Sat, Jul 28, 2007 at 12:03:33PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am facing problem in configuring D-channel for TE120P card.I did the
following things
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Hi,
I am facing some issues while using MixMonitor and StopMonitor. My
extensions logic is attached below:
exten = s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b)
exten = s,2,Dial(SIP/101,13)
exten = s,3,StopMonitor()
exten = s,4,NoOp(Dial Status: ${DIALSTATUS})
exten =
Hi Guy,. you should at least put a subject any way follow this link
http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530
Subject: [asterisk-users] (no subject) Hi, please help me in developing
Hi,
please help me in developing and reading Text through IVR application
using asterisk.
can any one help me at highlevel on this, other than using SPANDSP
application.
Regards
K.Rajesh.
_
Tried the new MSN Messenger? Its
6:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
___|
| || | |
Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+gotchas
iax.conf The new threading model is great, but the default of 10 threads is
way too low. Symptoms include total loss of audio until the channel is hung
up.
- in general section,
.
Bradiceanu
Sent: Wednesday, May 30, 2007 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g
otchas
iax.conf
The new threading model
Hello,
Did someone have a solution for a line fax detection for outgoing call
For exemple
I call number 0123456789
- if it is a fax then redirect to extension A
- if it is a line then redirect to exention B
whats ia want its somthing like AMD application that i use for the
answering machine .
Hello ,
iam having 6 asterik cards on three different servers
I am using asterisk 1.2.14 with zaptel 1.2.12 on fedora core 5 (2.6.17.1).
now every 3 days i need to rmmod/modprobe wctdm driver to detect the call.
callers wont get the IVR prompt. after rmmod and modprobe the wctdm ..it
works fine.
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You seem to have misplaced your message/comment/question.
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hi every body,
i m new to this mail list, and also with asterisk IPBX,
i havr digium TE110P card, can someone till me if he has an particular
experience with this card, kind of bugs, problems...
kind regards
Younss
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All:
I am looking to move cell phone providers. I have acquired the new cell
phone and LOVE my new number but want to keep the old number as well. The
new provider only will allow me to use one number or the other. They will
port the old number if I want, but will keep my new number if I ask
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and
7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
I was
Todd- Asterisk wrote:
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and 7
extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I
need the Sangoma A20200 or even the A20200D (Echo
At 05:23 AM 12/14/2006, you wrote:
Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
When I started down this path I choose the TDM04 and have always had
occasional echo issues, not bad and not often, but it annoys the wife
and
Subject: [asterisk-users] (no subject)
Hello everyone! I'm planning on setting up a new system shortly and can't
pick the right card... We will have 2 or 3 lines coming in and 7
extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I
need the Sangoma A20200 or even the A20200D
the sangoma A200 with echo cancelation and I have been
real happy.
- Original Message -
From: Todd- Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 14, 2006 3:23 PM
Subject: [asterisk
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use the
Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls
can I handle simultaneously.
I want to use the cards with the following Configurations:
Intel® Xeon 3.00GHz/800MHz, 2M Processor
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even
when it was recording 50% of the calls.
PaulH
On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote:
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use
the Digium product THREE
Here's a question maybe someone can help me with:
My extension looks like this:
exten = 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3
When I try this extension, the following output appears in the CLI:
Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored
i am sure this came up before
but all my searches are not resulting in anything usefull
trying to setup a grandstream phone
to connect to an asterisk server
now i am outside the network (home)
on my side
settings on the phone seem to be correct
id and password, astersik server ip, port
in
Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various tones according to different country standards
would it be possible to
Hi All
I would greatly appreciate some advice or some direction as
to where to go next.
I have a provider passing me incoming calls via my Session
Border Controller.
I am able to pass them calls fine but coming in fails with a
407 Authentication Fail error.
In my sip.conf I have
You might want to repost it with a subject or you miss a lot of people seeing or opening it up.
-- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED]
Hi All
I would greatly appreciate some advice or some direction as to where to go next.
I have a provider
Date: Thu, 19 Oct 2006 09:30:38 -0500
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk hangs up on incoming analog
calls after a while
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
I have two questions. First I am running a t400p with
three fxo ports signalling fxs (inbound CO lines). I have six polycom 501's. The
problem is the amount of time the call setup takes. I have done this with Mitel
phones before with a t-1 and had the same problem. My customer always complains
I am going to reply inline as you asked
many questions
I have two questions.
Sure, you do!!
First I am running a t400p with three fxo
ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem
is the amount of time the call setup takes. I have done this
Hi,
Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE,
India Pakistan.
Thank you.
John
mail2web - Check your email from the web at
http://mail2web.com/ .
___
[EMAIL PROTECTED] wrote:
Hi,
Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE,
India Pakistan.
This is a -biz question, not -users.
Also, do you realize how bad it makes you look that you can't even
bother to put a subject on your mail?
B.
--
This message has been
Hi,Is there somewhere a sample configuration for asterisk as gateway (pri - isdn)Thanks
Stay in the know. Pulse on the new Yahoo.com. Check it out.
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Best RegardsSony V. Shandy
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i have been wondering about how the useragents work since a month or two. i have tried every document possible...could not find the answer.
if anyone could tell me about the useragents, how they work, what are the factors that are considered while choosing a UA, what makes a particular UA best,
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Sent RTP packet to 293.67.65.3:43294 (type 18, seq 59050, ts
697456, len 2)
Got RTP packet from 21.98.11.200:58654 (type 18, seq 6246,
ts 3559220, len 20)
ANY ONE KNOWS WHAT THIS rtp DEBUD MEANS
THANKS
*
No employee
Dear
I am using trixbox,I want ot disable and enable voicemail
from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was
working successfully
Database put AMPUSER/9990999 voicemail default
And
Database put AMPUSER.9990999 voicemail disables
But at
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Hey everybody,
Is it alright to run two TDM400s on the same machine?
If it is, how would one differentiate between the channels on each card?
So, if Im running strait FXS and my first card is fxsks 1-4, would the
second be fxsks 5-8? Would there be any interrupt problems?
Any help
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
!-- /* Style Definitions */
-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Ninneman, Tj
Enviado el: Miercoles, 28 de Junio de 2006 12:54 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] (no subject)
Hey everybody,
Is it alright to run two TDM400s on the same machine? If it is, how would
@lists.digium.com
Asunto: [Asterisk-Users] (no subject)
Hey everybody,
Is it alright to run two TDM400s on the same machine? If it is, how would
one differentiate between the channels on each card? So, if I'm running
strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8?
Would
Hi,
I have the same problem with the queue configuration
When I receive 2 calls only 1 phone ring even if more agent's phone are free.
The second call will go to an other agent only if the first call is pickup.
Somebody have a solution ?This is my config file :Queue.conf[general]
;
I have a cisco VPN from router to router over a Data T-1.
The ping times are consistently 32ms with random ping responses of 295ms -408ms
about every 30 secs to a minute, I have jitter buffer enabled. The connection
goes like this Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B
Good day, Hi! i've finish up setting * for my company and they are working
reallly great, but i notice when i try to call to mobile phone, i can see the
zap channels is bridging successfully but i hear nothing except for a long
dialtone like tone, but calling to a regular pots line is working
[EMAIL PROTECTED] could be a better start for beginners (but beware, the
installation CD will format your HD without asking).
http://asteriskathome.sourceforge.net/
On Tue, 2006-04-25 at 10:47 +0800, rommel malana wrote:
Goodday,
I'm an opensource fanatic and I have already installed
--- rommel malana [EMAIL PROTECTED] wrote:
Goodday,
I'm an opensource fanatic and I have already
installed asterisk in my
mandriva linux. Actually, I'm also planning to
install the asterisk
management portal for GUI of asterisk. If anyone
could help me guide
in installing this. Thanks
Goodday,
I'm an opensource fanatic and I have already installed asterisk in my
mandriva linux. Actually, I'm also planning to install the asterisk
management portal for GUI of asterisk. If anyone could help me guide
in installing this. Thanks a mill for the help..
-Rommel-
Please make sure to write a subject line.
Thank You
On 4/24/06, rommel malana [EMAIL PROTECTED] wrote:
Goodday,
I'm an opensource fanatic and I have already installed asterisk in my
mandriva linux. Actually, I'm also planning to install the asterisk
management portal for GUI of asterisk. If
Currently, compiling asterisk on an Itanium fails with the GSM codec.
All I could find on Google was a hack to basically remove GSM from the
build which is not an option for some. This patch will allow it to
compile and seems to work perfectly.
Thanks,
Steve Totaro
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I have had the exact same problem last week. I have not yet solved it.
So instead I am using ooh323, but would prefer to use oh323. Can anyone
help?
I'm glad that I'm not the only one :))
Hopefully we'll find solution to this
Hi,
I'm using IPSwitchboard v 1.8.10, a sort of Operator Panel, to monitor my
Asterisk's extensions.
Recently I noticed that on the official site
(http://ipswitchboard.thorben.dk/), where I downloaded the software some weeks
ago, this project is no longer supported.
Is there anyone that can
Hi everybody. Yesterday I fix typo in spinlock.h and compiled zaptel. But today I have problems with soft phones. I tried to recompile zaptel and it showed errors again. So I don't understand what now it needs.
Brings words and photos together (easily) with
PhotoMail - it's free and works
Does anyone have a DISA alternative? I currently use the
line:
exten =
s,16,DISA(no-password|from-internal)
however that just drops a user at a dial tone, what I would
like to do is prompt user for number to dial, followed by the # key, and then
have asterisk dial out. Can this be
-0800
From: Martin Joseph [EMAIL PROTECTED]
Subject: [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not
getting?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US
Hi, to all,
i am new in the list and i am interest to deploy a sistem with asterisk i
have a PC with a Suse Linux 8.2 and also i have Dialogic VFX card with 4
analog port.
From where a can get Dialogic Driver for linux.
From ware a mast beging to resolve the problem the project to implement
luck
Jose Manuel Cortes David
X Semestre Ingenieria Electronica
PONTIFICIA UNIVERSIDAD JAVERIANA
De: [EMAIL PROTECTED] en nombre de Savvas Gavriel
Enviado el: Mié 15/03/2006 15:12
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] (no subject)
Hi
AFIAK, they can't - we would like to do the same thing, but it's not
possible with patching the source.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 10-Mar-06, at 7:56 PM, btb wrote:
can the
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can the default voicemail folders (old, work, friends, etc.) be
changed? for example, i'd like to configure asterisk so that there
are only folders called friends and old.
thanks
-ben
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HiDoes someone have a better sql query for selecting the provider used by LCDial application than the one proposed in the tgz ? It's far from working well with most of price lists.I tried to tweak it somehow with more or less success.Regards, Michel -- Michel Luczak[EMAIL
hi all,
see i have problem with PC(any sip phone which registered to
fwd.pulver.com) to phone(my zap where it has been registered by
modifying sip.conf)
my zap detects RBT but i am not able to listen to the voice,this
happened when i
tried with ECHO of fwd.pulver.com
i dont know wat to do plz
Hi,
This is test mail.
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asterisk-users@lists.digium.com
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Mauricio,
Yes it is. However I would not use analog phones.
Your cheapest option would be to use softphones on a computer. If you wanted to
use physical phones you have a few options.
1)Get two ATA's (device that you plug in to the LAN
on your end and by your friend to the internet). This is
Hi all,
I am testing my hands on asterisk , but got stuck. Let me tell you i am
only using its VOIP functionlities
I have registered the asterisk server at a remote proxy server. My clients
registered at asterisk server can make outgoing calls , but the calls made
from outside is not
Hi to all,
the following is the last thing we see from Asterisk befor it crashes:
$$$ find_chan_holded: No channel found for oad:017670014533 dad:7051538
-- ch-state CONNECTED, bc-holded 0
$$$ Bchan deActivated addr 51400101
-- cause 16
I SEND:RELEASE port:1 pid:88 mode:TE addr:51400101
See
http://www.iaxtel.com/setup.html
2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]:
Dear Sir
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk.
Thanks
.
- Original Message -
From: Branko Samardzic [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, December 02, 2005 10:43 AM
Subject: [Asterisk-Users] IAX trunking frequency parameter works only
oninitiator side
Hi,
I am experimenting with trunkfreq parameter.
When
-users@lists.digium.com
Sent: Friday, December 02, 2005 10:43 AM
Subject: [Asterisk-Users] IAX trunking frequency parameter works only
oninitiator side
Hi,
I am experimenting with trunkfreq parameter.
When it is 20ms I can see both parties in IAX session sending IAX frames
every 20ms.
When I modify
Dear Sir
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk.
Thanks
Nirukshitha Gamage
--
This e-mail and any attachments are intended for the above named
When dialing in after hours callers get to use the directory. I know
that if you put h or H with a Dial() command you get the behavior
of being able to terminate a call by pressing *. However, nowhere in
the entire extensions.conf does there appear the h or H option, so
I know it is not
Hi,
I seek solution for hotel management and billing solution. but I do not
know which to choose between Astbill or Asterbill ? if you have council.
Thx
David
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I have a Wildcard TDM400P card being used with Asterisk. For some
reason, incoming PSTN calls are getting delayed before they ring through
on the Asterisk PBX to an extension. The calling party hears an initial
ring tone and then a click noise, at which point it will then actually
starts to ring
I am having the same problem, but on both PSTN and a Voicepulse Connect
IAX line. PSTN rings clicks dead air, then rings and connects, IAX just
clicks, has dead air, rings and connects. Don't have a clue on how to
fix it though.
Greg
Roger Johnsen wrote:
I have a Wildcard TDM400P card
My Polycom IP301 hangs on
Processing Cfg...
Here is the boot log:
0930155446|so
|4|00|-- Initial log entry --
0930155446|so |4|00|+++
Note that bootrom log times are in GMT +++
0930155446|wdog
|4|00|Initial log entry
0930155446|cfg
|4|00|Initial log entry
Jonathan k. Creasy wrote:
0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded,
getting next file.
Any ideas? I attached the config files, I got them from somewhere else.
The phone isn't finding the config file as the above log entry shows.
The config file consists of the
When I receive voicemail notify via e-mail I would like receive not the
phone-number, but the sender name. Where can I configure this and how?
Is it possible to have some example?
Thank
Luca
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Hi. I
am using the Flash operator panel 0.24 and it works, but I don't see
the voicemail icon when I have incoming voicemail.
In the op_buttons.cfg I have the following setup:
[SIP/100] Position=2
Label="Office tel. 1" Extension=100
Icon=1 Mailbox=100 I've tried
to google on
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102 etc. but when i truy to go outside with the 9 before the call rings
in the first extensions (100). this is a asterisk problem? or a pbx
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from
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