Which has existed, in one form or another, for years. I was using a
voice enabled faxmodem a decade ago to answer my phone. The software
that came with it (don't remember the name, but WinFax also does/did
this) even allowed for a simple IVR, for mailbox selection and whatnot.
The only things
A little off topic (sorry..:) ) but anyone know what format Cisco phones
use for their contact dirctories. I want to set up my contact lists on
the phone, and cannot seem to get any info on it. I am working with a
7970 on Asterisk 1.4.8.
7940 and 7960 use this format of XML file (probably the
In fact, my questions are more about usage than about technical background.
For instance, I doubt a user will log his system off when leaving : some
don't even turn their PC off.
Does anyone has an experience to share about that ?
When I tried it, when a user login at a phone, it replaced
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
You can try my IAX2 softphone for windows :
http://www.marccharbonneau.com/asterisk/mediaxphone.php
Hope it fits your need
___
Can anyone post a sample of whats needed in iax.conf for an IAX UA to be
able to make and receive calls?
[7011]
type=friend
secret=S0m3S3cur3P4ssw0rd
qualify=no
notransfer=yes
[EMAIL PROTECTED]
host=dynamic
disallow=all
allow=ulaw,alaw,gsm
context=from-internal
callerid=Marc Charbonneau 7011
So I am looking for a softphone thats really simple to setup and as
foolproof as possible..
If SIP is likely to be problematic to setup then I have no problem
getting him to use IAX but will need suggestions of which IAX softphone
to use and also how to configure it in the iax.conf (haven't
Could I rewrite this in Delphi instead?
I never used Delphi to write an AGI but I've seen a class in
FreePascal that you could probably use as a base :
http://www.automated.it/asterisk/fpc-agi.html
hth
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A. Yes, I have the cojones. He never mentioned what platform it was for.
We need something like this for Linux. I got all excited about it only
to be terribly disappointed when I unpacked it.
From the original announcement : It runs on any modern flavor of Windows.
It is not like if he said
First - vtiger is available for those who don't like the SugarCRM
licensing.
It's not a licensing complaint. At least that has not surfaced. It is more
that the
programmer does not seem to be comfortable with SugarCRM, MySQL and php.
Biggest compliant about sugar is - hard to configure,
need to preprocess
1) incoming call get caller id lookup some info in my db,
2) based on the result dispatch the call to the right operator
step 1 is ok I developped a small .php script that connect manager and
parse events, now I have to tell AAH do dispatch call to the right
operator
From your
Hi List,
I wanted to know if anyone knew of a way with asterisk to switch the voice
of a caller from male to female or vice versa.
http://www.lobstertech.com/code/voicechanger/
hth
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Any ideas on this?
Closest thing that comes to mind is FOP : http://www.asternic.org/
hth
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Anyone have an idea to re master centos,in other worlds I have an asterisk
on centos with all libraries and modules,how can I make it as an iso image
?
Have a look at Kickstart
hth
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Responsibility for answering the door is shared by the entire office. But A) noone wants
their phone to ring, there's a door chime) and B) noone specific will accept
responsibility for answering the door. So, we need a solution that follow I'm
answering the door now, these are the buttons I
On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote:
I looked at a call queue, but it didn't seem to work the way I want. Agents
need to log into the queue to get calls, seemingly. Of course, I only stopped
on the topic for a short period. with the meetme conference, anyone can answer
the door
Does anyone know how to configure a SIP phone to pass the mailbox number to
the voicemail service when dialing? I would like to press the message
waiting lamp and be prompted for my password instead of mailbox number.
Can this be passed in the set-up call or based on caller-id?
based on
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and then
the file will be played from start again.
I would like that the play of file is only stopped if the user has pressed the
key 3.
What for an command can
You mean compiling raw tar.gz or SRPMS? And where do you download
them from? Trixbox site or the original vendors' sites?
I just download the tarball from asterisk.org and compile it. Trixbox
is not a special version of Asterisk, it is just an easy way to
install Asterisk, FreePBX, FOP and a
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Asterisk recognises them as native uLaw files
From what I know, .ulaw
I also include a consideration from mine: I would happily use
Trixbox, because I did FreePBX setup once and it was a real pain, but
I'm very frightened by a few issues:
1) Trixbox Macho installation that installs everything without
asking. I, for example, would like to use software RAID (maybe
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
What Error ? it says DEBUG
This just tell you that the user
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
Just use an
How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.
the
exten = ,n,Queue(|t|||300)
exten = *,1,Macro(agent-add,,)
exten = **,1,Macro(agent-del,,)
So my question is , what means these one/two asteriks (*,**
).Maybe it is like priority.?
It means that to login as an agent on the queue you have to dial
* and
This is the error i got. I've grepped through all of my include/linux/
wanpipe_includes.h files i have on my server (there is actually a
couple of them), and replaced config.h with autoconf.h, but still i
get the same error. Looks like I'm unable to locate the include/linux/
wanpipe_includes.h
Ok, how can i do the transfer from the caller to $keys ?
Probably by using a goto :
http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#goto
hth
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All the variables here was my_var, it worked for GET VARIABLE but didn't for
SAYDIGITS and odbc connection. How can I SAYDIGITS of my_var or insert
my_var value into a db?
- What I need more to use WAIT FOR DIGIT? Because it didn't stop to wait for
digits.
- STDIN shoudn't get the result of READ
after the Dial has connected, I want the caller (on a SIP phone) to be
able to press keys in order to record call status. is this possible?
Have a look here :
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
applicationmap is what you are looking for
hth
The phone in question just prepended 010whatever to ALL phone numbers
dialled, which makes it pretty crappy to use with a line that does not
allow for network selection codes, or on lines that need a 0 for a
POTS line.
You could use it as an extension on Asterisk and strip that pre-pended
number
Hi - I just got a Sangoma A200 card with a single 2FXO module and
what appears to be an empty module. I put the card in my Dell GX260,
but the power light on the front of the box just blinks and won't
power up.
Maybe your card is not properly seated.
seems to have a lack of documentation, but
Thanks - that turned out to be the problem. Well- one of those solutions.
I removed the blank and swapped the FXO module to the other port. I don't
know if it was a bad port on the A200, but since I don't plan on using it, I
won't worry about it- just regret it in a year when I get a second FXO
I wish you all a Happy 2007 filled with an almost-bug-free,
full-of-nice-features Asterisk 1.4 :c)
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I am using the Background() function to ask for the extension, but the
extensions are in a different context. Is there a way to tell Background()
to look for the entered extensions in another context other than the
currently running one?
in that context you can do
include = other-context
hth
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown
Is there a way I can create a _NXX extension and insert 1 and areacode
when dialing?
exten = _NXX,1,Set(CALLERID(num)=6162997590)
exten = _NXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/1514${EXTEN})
replace 514 with your area code
hth
___
Below are a few errors in the script and on a google search, although I
found people with the same error, I didn't find a resolution.
Any thoughts on what is causing this error?
Any thoughts as to why the output is not showing on the CLI without doing a
debug?
snip
Content-type: text/html
I need a quality US 800 DID over IAX for my Asterisk server, preferably one
that doesn't cost the earth.
Any suggestions please?
Never used them but the rates seems ok : http://www.les.net/
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So I moved the Ethernet controller to IRQ 11 and the 'Unknown Device' followed!!
I would try moving the Digium card to another slot. Your Ethernet
controlled must be onboard and it share its IRQ with the slot where
the Digium board is.
hth
___
Now for some reason instead of giving me an error on the caller ID, it's
not mentioning the caller ID at all. Is there some explicit thing I
need to put in to get the caller ID?
callerid=asreceived
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camille*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudoincomingen
This should show something like this :
panoramix*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudofrom-pstn en
I've read http://www.voip-info.org/wiki-Asterisk+AGI+php but i can't
understand how to play sounds and read DTMF digits...
Have a look at this : http://phpagi.sourceforge.net/
hth
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exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten =
Just add a 9 in front, like this :
exten = _90[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _900.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _909XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Oups, pressed Send too fast, here is take 2
exten = _90[1-9].,1,Dial(IAX2/[EMAIL
I've been doing a lot of playing, and a lot of reading, and it seems people
are split as to whereas if they're running their favorite Linux distro and
asterisk or Trixbox. I'm getting closer to really looking at a production
environment and I'm just looking for any opinions. I'm really enjoying
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best
I can't risk spending a few thousand just to reach the
conclusion that Digium's PRI or BRI cards do not work
with a particular system's PCI-X slots/bus... Or, worse,
staying with a dead card / system board in my hands ! :-(
Anyone ?
I don't know about Digium cards, but I just
Does there seem to be a popular Linux distro folks use specifically for
Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros. In particular, I'm looking for a free, stable, well
supported distro that has a friendly community. Any advice appreciated.
CentOS
$25/month will buy you close to 50 hours of urban SIP termination,
it's down to half a cent in some of the big cities. Are you
going to average 50 hours on the phone each month? Some people
do, but most don't. (Otherwise Vonage could not even pretend it is
going to make money.)
You don't
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the
functionality to make the call out?
[from-pstn]
exten = s,1,Hangup
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How can i Processed the call Snooping, it my fifth Requesting and posting
to Users, Nobody replies it,,,
see http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy
hth
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I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card
connected to a POTS line and a phone set (physical extension). I've got
all incoming calls launching directly into an AGI script. I'd like to do
the same for the physical extension. In other words, when picking up the
hand
What I'd like to implement, ideally, is that once an incoming call is
transferred to a particular operator, the system also calls a manager
who can monitor silently.
I think you are looking for this :
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
if its a version 5 or higher, that wont be an option, but if its not,
give openwrt or ddwrt a try.
Actually, this is no longer true (at least for WRT54G), see
http://en.wikipedia.org/wiki/DD-WRT for the official list of supported
models
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Nov 30 00:19:06 ERROR[23493]: pbx.c:1382 ast_func_read: Function STAT not
registered
from http://voip-info.org/wiki/view/Asterisk+functions :
Functions in the below list are marked in red if they are only
available in version 1.4 and higher.
And STAT is marked in red so I guess you're not
I am attempting my first go at a simple AGI application using PHP (Getting
Asterisk to SAY PHONETIC ABC). I have dabbled with PHP but I am by no means
a professional standard developer.
Can't really say what is wrong with your code since I never did an AGI
in PHP without this class :
I use some custom scripts to do database lookups and rewrite CallerID
information. Everything works fine with regard to the CID name, however
my Cisco 7960 and Linksys SPA-942 phones do not display the calling
number. Instead, they display the called number. This makes the phone's
call return
for see if the card answer, what is the process ?
since your port is configured to be in the interne context, just add
this to this context
exten = s,1,Answer
exten = s,2,Playback(tt-monkeys)
exten = s,3,Hangup
watch the console and dial-in. if you get monkeys screaming at you, it worked !
I am using asterisk to receive call from a DID provider . In configured
everything in freepbx properly and its working . I forwarded incoming calls
from did to a certain extension . Now i tried calling from another sip
provider to this box , when i call from other provider to my DID number
then
Most of our customers have generic names like Hospital, so I need to
rewrite their caller ID name by looking up the number in a database on the
Asterisk server, and rewriting the name such as Reading Hospital so that
we know who's calling.
Any idea if this can be done with Asterisk, and how to
Thank you for the confirmation and the warning about disk space.
Now I need to decide between the Sangoma A20202 and the Digium TDM2411.
I'm leaning heavily toward the Sangoma card for the following reasons:
- It doesn't require a 12V power connector for the operation of FXS modules.
Maybe,
Is it possible to record outbound analog calls using an X100P?
I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users. I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
A200s with 4 FXO and 4
I believe that the problem really is fault of DNS lookups, but as I
should proceed for resolve that??
see the first point at
http://www.voip-info.org/wiki/view/Asterisk+administration
The best solution for now is probably to have a caching dns server on
your Asterisk box or in your LAN
Apologies.. we are using a sangom 4 port FXO card. It used to work
(or so the company claims that has the PBX), but they are saying it
stopped.. yet nothing has changed on the PBX system. I have verified
it IS picking up and then passing the call onto the ringgroup (hence
taking it out of the
Today we appear to have discovered our first bug. We have an extension
setup to followme by ringing that extension + an external cell #
(ringall). If nobody answers after 20 seconds the destination if no
answer is set to go to the extensions voicemail in the followme module.
The problem is it
After installing properly when opening in the webpage it is not parsing the
index.php for the AMP. My Database is MySQL.and web server is Apache 2.2.
Please let me know is this configuration problem or this is the problem with
Apache (Apache 2.2) .
The problem is probably that you didn't
I have a new client who has an existing Asterisk PABX and is looking
for us to install a TE110P for him, However he has a Dell SC420 and I
have never used one before.
I have had no problems with any other Dell servers which we use almost
exclusively.
Has anyone had any good/bad experiences
Then what would be a better solution?
Usually the IAX phone will play you a ring tone until the other end
answer. If you're phone doesn't do it, then it is a flaw in that
phone.
What phone is this ?
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...but I'll need to give the users a good mean to see
what's going on,
who is busy,
easy transfer with click here and there,
easy conference,
easy queue handler,
easy way to see/use many lines at the same time
is there any best console they can use?
Have a look at FOP : http://www.asternic.org/
Is the Wildcard X100P still supported? I have one sitting around that I
bought 3+ years ago and never used it. I need the functionality now.
Before I run off and buy something new, I'm curious if this will just
work.
It still works with the latest Zaptel (1.2.10)
I also have an old TDM400P
Thanks to all that replayed, I made like Mr Watkins told me, and my problem is
apparently solved, although, because of the usage of the syntax
VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played vm-intro and beep.
Is there a
Now I'm ready to begin playing with dial plans and am having a difficult
time getting started.
You may want to read the book :
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
That should help you
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You've got a very poor grasp on how things work. Please don't pretend to know
what you're talking about.
# netstat -apn | grep :80
tcp0 0 0.0.0.0:80 0.0.0.0:* LISTEN
782/httpd
tcp0 0 204.xxx.yyy.188:8080.xxx.yyy.167:58620
ESTABLISHED
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
You probably have some script that use the console to query something,
like the
Why is it running on port 1207?
because Asterisk is listening on port 4569 and when a connection comes
in, it as handed to another port so it can continue listening on port
4569. Otherwise you would only be handling 1 connection at a time.
Pretty basic networking stuff I think :c)
Thanks for the answer, but I don't buy it. There are currently 0
calls up on that bridge, while another connection which has calls up
on it is on Port 4569.. please try again. IAX2 is suppose to run on
ONLY one port.. this is why it is so nice for use in firewall
situations.
It doesn't change
On 10/16/06, Melcon Moraes [EMAIL PROTECTED] wrote:
OMG, please read more about network ports.
Could you tell me what is wrong with my explanation ?
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Ideal would be a headset audio+microphone with RJ11 4p female that we
could plug into the handset cable of any IP phone, or a converter
2xjack2,5mm female RJ11 4p female -which seems not to exist-.
What are you recommanding/using/installing in such case?
I don't know if it would work on any
Is it possible to connect a modem to a remote service through asterisk ?
Basicly to ilustrate : Accounting department need to connect with analog
modem to their bank to order some wire transfert.
Modem - Chanel Bank FXS - Asterisk - TDM2400 FXO - Modem in
remote site.
If you get it working
I'm looking to do this.
When a call comes in to an agent in a queue, pop a web page like this
http://www.mydomain.com/cgi-bin/script.cgi?did=952900
Where did is the number the caller dialed to reach the system in the
first place.
I know Hudlite can do this we caller ID, but the DID feature
I'm deploying an asterisk PBX for a Call Center and i was ordered to
check if the Customer Support Supervisor could intercept the calls so
they can check how they employees work with Asterisk.
have a look at these :
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
It'd be cool if someone wrote a syslog() dialplan application for Asterisk
*hint* *hint*
That could be usefull, but what is wrong with : System(logger Asterisk
can use syslog) ?
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Is there a more elegant way to tell it to answer/not answer on command?
Put your Zap line in a context that do just this :
s,1,Hangup()
hth
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Just wondering if there are any IAX phones worthy of the name phone out
there -- looking for hard phones, but I suppose a Linux-based softphone
wouldn't, you know, hurt. ;-)
Idefisk looks pretty nice and there is a Linux version :
http://www.asteriskguru.com/idefisk/
There is also iaxcomm :
What I would like to do is in my flash hook dialplan code to ass
something like Hangup(SIP/100-fe65), but where can I get that
SIP/100-fe65 ? Is there a variable set with this information available
in the dialplan ?
${CHANNEL}
have a look here :
A Cisco (used) 7940 that's loaded with MGCP and I want to load SIP so it'll
work on my Asterisk box (outside of the FXO FXS modules on the TDM card in
the Asterisk server, I only run SIP on the hardphones).
I don't know the phone's password (sound familiar?). - Have tried
everything, cisco,
I need to pass modem calls through a TDM400 card. Conecting the modem to
the FXS port (ZAP/1), it should be put through the FXO port (ZAP/4)
directly.
According to Digium, Fax calls (and modem calls) are not supported on
the TDM400 or TDM2400. They are designed for voice only. If you get it
to
At some point in my dial plan, I need to find out the length of a sound
file in seconds (to weed out things that are way too short)
the record application doesn't seem to have any facilities to do that.
any ideas ?
use sox beep.wav -e stat and parse the output
man is your friend
google also
We have a problem in configuring Sangoma A104. We want the 2 ports to be
configured as E1 and the 2 ports as T1.
If I'm not mistaken, you can't do that with the A104D, that's why they
sold me 2 x A102 for the same price as a A104. Better check with
Sangoma.
hth
keeping track of the confno is easy since I created it,
but I don't know how to determine the user number of the last person that
joined the conference.
Is there a way to store this in a variable before they join the conference?
Or perhaps a way to detect the last user to join the conferences
Does ANYONE have any clues?
Only played with 7940 and 7960, but I will try to help since nobody
comes forward
loadInformationSIP70.8-0-3S/loadInformation
Shouldn't that be something like P0S3-08-2-00 ?
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I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
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On 8/23/06, Infobox Peru [EMAIL PROTECTED] wrote:
maybe you could make it with PHP and its driver for Oracle.
For PHP have a look here : http://phpagi.sourceforge.net/
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Can someone recommend a good text to speech engine that is usable by Asterisk?
I have tried the Festival one and it just doesn't cut it for commercial
applications.
We are willing to pay for a good one that works. Anyone tried the ATT
All I can find for Flite is for AAH, does it work as well with plain
Asterisk? Is the setup the same?
Never tried it, but it should be the same.
Have a look here : http://dialogpalette.sourceforge.net/extras.html
hth
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I've installed asterisk and astbill according with all recommendation
(mysql5, drupal included with astbill, php, apache2...).
When I write http://server_adress/astbill, I get a white screen page.
Browser doesn´t give me an error page, it just a white screen page.
you have to enable it in php
The problem a number of people are not entering the pin fast enough
,they are not given enough time to enter the PIN( I assume this is a
mailbox number)
looking at all the doc is seems everything is configurable, can some
one point me in the right direction of where to start looking?
check
Can anyone direct me to where I might find examples of handling
interactive input from a phone using PHP and AGI. I want to have someone
dial an extension and then have the system request input from the user,
take that input and put it into a database.
Start here :
#!/usr/bin/php -q
?php
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-say_digits(62410);
$cid = $agi-get_variable(dir);
$agi-say_digits($cid);
?
I'm getting this error:
parse error, unexpected '=' on line 6
I don't know why you're getting this error, it parse
Picture this:
Exten = 100 #My Phone
Exten = 200 #MythPhone
Call comes in. Dialplan calls both extensions.
MythPhone is an add-on for MythTV,so when i receive a call,the CallerID
is flashed up on my TV.
I want to add another MythPhone to my other MythTV box upstairs.
Do i have to make a third
Hi everyone,
I know that functions like set_variable and get_variable (using php with
phpagi) only apply to the channel variable. What I need to do is reset a
global variable I have in our system. I have a script that is going to
determine when this will happen, but I just have to make it
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