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it is possible to use musiconhold. i added
exten => s,5,SetMusicOnHold,default
exten => s,7,dial,SIP/michael&SIP/frank|15|m
to the extension conf.
the logfile looks good, i think
-- Called michael
-- Called joern
-- Started music on hold, class 'default', on CAPI[contr1/]
-- SIP/192.168.10
For those of you running OSX, a new h323 client was released. Haven't
set up h323 yet, so I can't vouch for it.
http://xmeeting.sourceforge.net/
--Mike
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hi!
I wanna do some arithmatic operations (addition and substraction -integer
operation) inside extensions.conf. Is there a simple way to do this. If I do
yy = ${xx} + 1 // say "xx" is initialized to '0'
the resulting "yy" will show
"0 + 1"
Obiviously not the result I need. Any h
Hi Asterisk-Users
I’ve been reading about
the Asterisk project (all that I could get my hands on J ). It sound to good to be true. But I’ve got some
questions which I haven’t found a answer to anywhere :
1) Can I use Asterisk as a Call
Manager using MGCP protocol or H.323
Hi everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that whe
Hey all,
I have a E1 setup with a E400P digium card. Everything works just great
except for the callerid. When i make an outgoing call via the E1 to a
hardphone somewhere, all i get is "private number". In my zapata.conf
however , i have defined the following:
context=localE1
group = 1
channel=1-
Is this me or what?
-- Playing 'demo-congrats'
-- Executing MeetMe("H323:996", "") in new stack
-- Playing 'conf-getconfno'
== Parsing '/etc/asterisk/meetme.conf': Found
WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open
pseudo channel
-- Playing 'conf-inval
I'm looking at getting the Dev light applications from digium
and I have 2 Createive Labs voip blasters. The voip blaster supports the
G.723.1 codec. After looking at Gnome meeting it does not talk unless you have a
quicknet card for it. Can I make calls using asterisk and the digium cards to the
Newbie question, please excuse me for this
one. If an admin adds and extension in the voicemail.conf file will
asterisks read from .conf files dynamically? Or does the asterisks daemon
need to be restarted? I guess this question pertains to all .conf
files. Also is there support for MySQ
Hi,
I am a new to Asterisk. I am looking for a cheap solution for PC-to-Phone
call serving one or two cities. Can anyone provide me with pointers to
architecture documents/other documents from where I can start. I am not
new to VoIP.
Regards,
Deepak Mittal
-
I got asterisk set up so that it doesn't take over
the sound devices now. Thanks to the list for that
:)
Now my problem is getting people from outside my
router to be able to connect / use asterisk to
leave me voice mail. I'm using SIP and trying to
get a friend's Windows Messenger to work with i
ðÏÞÅÍÕ ÂÙ É ÎÅÔ? ÷ÏÐÒÏÓ ÔÏÌØËÏ × ÔÏÍ, ËÁË Õ ÔÅÂÑ ÂÕÄÅÔ ÓÏÅÄÉÎÑÔØÓÑ ÐÁÎÁÓÏÎÉË
Ó ÁÓÔÅÒÉÓËÏÍ.
Date: Wed, 30 Jul 2003 20:06:17 +0400
From: Pavel Zheltouhov <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk,ata186 and Panasonic TD1232
Reply-To: [EMAIL PROTECTED]
I have Pan
Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls...
This looks rather interesting. They also have an IP phone which is
probably low cost, but it seems to only support G.723. Has anyone used any of
these products?
http://www.nicstel.com/2001/e_3023w.html
http://www.nicstel.com/2001/e_products02.htm
hello all
well while trying to make a call from gnophjone registered with IAXTEL to
another phone registered with iaxtel, we get disconnected everytime a call
is made, and the following message log is generated
17006383019 is a phone number
Using: [EMAIL PROTECTED]:5036/[EMAIL PROTECTED]
Trying
I have problem with a TDM40B installation.
When i modprobe wcfxs the error i get is the
following:
/lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No
such device
Hint: insmod errors can be caused by incorrect module
parameters, including invalid IO or IRQ parameters.
You may find more inf
Is it this maybe?
Communication controller: Tiger Jet Network Inc. Model
300 128k
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Hello,
I made install.
Why I am getting this.
My linux is Debian.
--
Hi
Looks like you did not do a make install after compiling the drivers, and
it is still loading the stock kernel ixj.
Please try doing a make install in the ixj-x.x.x source directory.
Hope that helps
On Tue,
Dear All,
I am going to deploy a VOIP
network here in Canada with nodes all over town. This is for long distance
services and hence would need a good reliable solution.
I have looked into * and am
very interested in it with all the value-added features as well as its
capability to do H323
Hello,
Can Asterisk perform as a H323 Gatekeeper?
Here is my scenario:
I have a customer that has a calling card program that will be
transmitted as VOIP from a Cisco 5300 in Hong Kong and terminated here
in North America. The catch is that, the termination is being handled
by a third party co
Hi,
Checked out latest CVS and no sound from Playback, Background, MOH or
bridged channels. mpg123 is active but no sound.
Master
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> Brian> Hello, I resolved my echo issue using grandstream/estara
> etc etc Brian> sip phones and wcfxo interfaces from digium. I
> swapped out my Brian> via kt400 based msi kt4vl motherboard for an
> asus p4pe? i845? Brian> based motherboard and the echo has
> completly gone away along Brian> wit
how do I prevent people from calling as soon as I restart the * server ? cos' this
will result (I assume)
in pri channels getting blocked. Because of those few calls that's taken during
restart results in
those few pri channels not to get properly restarted. I need something like 1~2
minutes
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7083533
I try to install asterisk on mandrake 9.
when I run "make" as root
I get "error 1"
any clues?
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Hi guys,
I am having a problem that I can't find an answer on digium and or the list.
When a call covers to vm, and starts to record the message I get the
following on the console. I am running redhat 9, last nites source, with
one X100P and a TDM400 with three extensions. I have tried to find an
Hi,
First at alll, I beg your pardon because maybe I explained bad my questions
(because my low level english)
I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2
compiled and installed.
I have modules alsa 0.9.8 compiled and installed
My PC has an audio card ac97 chi
Hi,
Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that covers
needs in that area: scratch number generation, call
termination via least-cost route (i.e. multiple
termination p
Hi all
How can I make * ring one phone then if no answer
Go to a different extension ??
Any help always appreciated
Regards Mick
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Dear All,
I have had a
problem that I have posted before, the asterisk kept crashing on me. I have
thought that may be before the problem is resolved, I could try to implement a
cronjob to run /usr/sbin/safe_asterisk, and if Asterisk is not running at that
time, it will start it automati
Hello,
My clients usually work the regular 8 - 5 day, however they
would like to have control of the night time context.
Is there any way, say a receptionist, can dial a 4 digit extension, to
toggle on/off the night time context?
Thanks in advance,
Brent
___
We are new in Asterisk - I was wondering if someone can recommend a good
phone sets to use with Asterisk in office environment. We need about 20
sets.
Also - What can we use for the receptionist phone?
Thanks,
Aram Ter-Martirosyan
Dear All,
I recently came across
DynEXTENdb, a way to be able to include thousands of Extensions (routes). In my
application which is VOIP, we need to include more than 50,000 area codes due to
the USA LATA routing, and there is simply no way to do that with
extensions.conf. The way DynEX
I am a new asterisk user and I love what I see so far.
I have a question about distinctive ring though.
In my situation, we have 1 phone number for voice calls and one for faxes.
They share the same line, and right now I use vgetty with mgetty+sendfax and
VOCP to deal with calls and faxes.
Vgetty d
bkw,
I realised that I was running asterisk with just asterisk no cli options
changed it to safe_asterisk any my problem went away, so it might just be that it
doesn't want to work in asterisk, just safe_asterisk
when I some free time I'll get a coredump since there are no real informative debug
Hi all.
If I want to use the * only as a GW to PSTN and allow only one external
proxy to place calls. how is the smartest way to do this ?
I dont want "the world" to be able to do invites only a specific IP,
in this case my proxy that handles all the users.
/Mike
Hi guys,
Has anyone played around/got it to work app_prepaid.c?
(http://www.voip-info.org/wiki-Asterisk+callingcard)
With what data do you populate the database with cards, providers,
tariffs, tariffrates etc.. (format) to make it work. What is the
meaning/purpose of each table/field?
I am gettin
Dialing in from the pstn to sip phones (x-lite softphone on winders and
a grandstream handytone-286 ata), I hear the sip phone ring a few times,
I ran into the same thing with Cisco 7960. Looks like the logic in the
sip channel has changed recently.
Add a ",r" to the end of your Dial statements i
When my snom200 receives an inbound SIP external sip call,
it somehow rejects the call and with a busy tone. The debug shows the
following error:
channel.c:1142 ast_read: Exception flag set on
'SIP/sipphone-7796', but no exception handler
what does this mean and how can I debug
Has
anyone had any luck using a 7910 with SIP image.
Some
information I found says 7910 is skinny only, other info suggests the 7910 may
take the 7960 sip image.
Can
anyone offer their experience ?
Cheers
Peter
Hi!
I have a little big problem here. I have an gateway(asterisk,working as a H.323
- SIP gateway) conected to a gatekeeper (two different servers), and also a
gateway (cisco - PSTN) conected to the same gatekeeper. When I make a call from
the gateway(cisco) to a sip phone, the phone rings, but wh
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I am getting ready to do my first build on this product. It's just for use as an overglorified answering machine right now and I will most likely play with some of the SIP functionality.
My big question though, is how much disk space do messages take up on the system? Are there any published metr
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 gatewa
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 o
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN->PSTN calls get this error...
-Greg
<--
Dear Steve,
We have installed Asterisk with Digium card TE110P , install MFC R2 connect to
PSTN (indonesia) using DIG13 MFCR2 siemens EWSD, Germany.
asterisk working normaly, outgoing call ok, incoming call ok. but in central
office /PSTN having SLA(service level alarm). If It happend, all chann
I heard something about the agents.conf file in the asterisk pbx.. I would
love to have a tutorial or someone that will help me doing this.. it's not
working out with her
Can anyone help ? it's getting frustrating with teaching the agents to
logoff the queue everytime.. or even teaching the superv
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 thing
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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.
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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: ${DIALSTATUS})
exten => s,5,Goto(sss-${DIA
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,
Niki
Hello,
I have problem after killall -9 asterisk
and asterisk -f
Asterisk stops to send to DNS resolving of trunks
Regards
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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 thinking
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 t
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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In the setup tutorial @
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
it states the potential issue regarding setting up UniqueID
as the primary key, but doesn't state how to rectify this?
What is the proper way to make sure this is done right?
Also, has anyone buil
Hi :
asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main
configured files are:
extensions.conf:
[from-pstn]
exten => 9711315,1,Dial(SIP/3000,30)
exten => 9711315,2,VoiceMail([EMAIL PROTECTED])
exten => 9711315,3,PlayBack(vm-goodbye)
exten => 9711315,4,HangUp()
sip.conf:
[3000]
Hi
I m making a call from one asterisk server to an asterisk client
The call gets completed but I want it to send dtmf signals
The dialplan I have made for this is like
exten => 205,1,Answer
exten => 205,n,Wait(15)
exten => 205,n,Playback(dtmf-1)
exten => 205,n,Wait(20)
but it does not send any
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
loadzone = in
defaultzone
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using
CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I
try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
I just
Hi All,
I have one doubt, suppose we have conference between 3
users (PCM
companded voice channels) then we add the streams together with scaling but
data which a user can receive will include his own voice information also
or i think we should substract his info. from the
V 1.4
When I do a "show channels" I get the following.
CLI> show channels
Channel Location State Application(Data)
SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up
Page(&Local/[EMAIL PROTECTED]&Local/71
SIP/7110-afd286e0[EMAIL PROTECTED]:2Up
Page(&
ko gui nua
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I have to develop a VoIP application. I need to know how to use Java APIs to
communicate to my client application with asterisk.
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Hi
I use dial with music on hold command
exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , t
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All,
I want to know if it's possible to create a log file per context? and each
time a context is restarted a ne x log file is created.
regards
Mickael
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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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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 .
Need some help with IAX trunking.
I've got six systems:
AsteriskM (main)
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Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5
AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other A
Hi,
That model HP or Dell server that I recommend for a TE412P card for about 200
users?
Thank you very much.
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Please descard me from the asterisk users list...thanks
(Abu Nasar Mahmud)
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Hello,
I'm looking for some advice on securing Asterisk.
Recently my servers been under several brute-force SIP attacks.
I have several remote sites, as well as many roaming users, who may have
PC softclients and/or SIP based hardphones.
My first step will be to strengthen the password
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Good morning all..
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have the
S.NASROLLAHI
hi
i am a new member
i want to learn what is TOS and LOG command in the access list and
what are they doing?
what is their advantage ?
when i should use them?
thank u
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Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have no idea how long
that error has been there, but I'm just cur
hello,i'm a naw asterisk user i've configured my 2 sip phones and they can place calls .i,ve also fxo card and i've configured channel ;now it's possible to recieve analog calls with my sip phone but i want to make call with my sip phone to analog it's possible?
when i dial a number my sip phone a
Hi friends !
Cvan anybody help me to configure asterisk with ser so that I can share the
load of the asterisk with ser server. I have tried it but my asterisk is not
showing registrations of the useragent, as given in the asterisk
wiki/asterisk+at+large. I don`t know what is the problem, but c
When setting up the que's do you have to add the que to the context?
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Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten => i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten => s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
exten => s,2,G
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Hi,
my setup:
Client: Win/linux client running x-lite or linphone
Server: debian running asterisk
on connect, incomming works well but outgoing to POTS has a lot of bad
sound (no, the mic is ok, using logitec usb headset). to ensure proper
work, tried normal p2p, worx well
the sound is nearly una
Hi All,
I recently upgraded from a very old stable to HEAD. For some reason,
incoming callers don't hear ring tones when calling in. Everything else
is working fine. Where should I look for a fix?
ISDN --> X100P --> asterisk --> sipphones.
Thanks
Johan
Hi all,
I would like to study the asterisk source code(Program). I dont' know from which file
i've to start reading the code. can anyone helpme.
Regards
Shan.
Hello again!
Just wondering if any one else has had a problem with stop and starting
asterisk?!? If I do it say 5/6times without restarting the computer then it
crashes. This doesn't seem normal to me, could this be because I'm running
fedora core 2? I know there's problems with using fedora to do
Hello,
does anybody have any experience with CNAME
resource records in e164 zones.
Example:
e164.arpa zone
3.3.0.3.7.2.7.2.2.5.3.e164.arpa. IN CNAME
3.3.0.3.7.2.7.2.2.5.3.e164.lu.
e164.lu zone
3.3.0.3.7.2.7.2.2.5.3.e164.lu. IN NAPTR 100 10 "u"
"E2U+SIP"
"!^\\+35227273033(.*)$!sip:[EMAIL PROTE
Hello,
I have X101P card.
But it seems to be dead. Always
app_dial.c:803
dial_exec: Unable to create channel of type 'Zap' (cause 0)
I've add the
line: exten => 999,1,Dial(Zap/1). But calling to 999 show the same
error.Zap show channel, lspci etc show everything is normal.
Could you tell
Hello
I setup Mediatrix 1124, I am able to make incoming call but unable to make
outgoing call. When ever I tried it just gave me a beep sound.
I appreciate any help on this.
Thanks
Deepak Malhotra
This message was sent us
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
[EMAIL PROTECTED]:X:[EMAIL PROTECTED]/1234
[sip.broadvoice.com]
ty
Does anyone now an Asterisk consultant in Atlanta?
Bobby
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Hi Roger!
the databse i have created has all the immaginable permission.if i try to
access it with dbtools with the
username and the password of asterisk from another pc inside the network, i can
see the table
and i can read/write. Only * can't write into the table. So,cecking the conf
file and
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