On Wed, Feb 18, 2009 at 3:51 AM, David Backeberg dbackeb...@gmail.com wrote:
As for actually putting delays and pressing the right buttons, you're
on your own. You would need to write a custom AGI script specific to
your IVR, and call it from your call file, which you then put in a
bash loop.
On Tue, 17 Feb 2009, Mark Michelson wrote:
The purpose of exposing these values is to allow for an administrator to
use these for any purpose he may desire.
An example would be really great :)
I am confused because these values are exported just before the call is
connected and I am
Hi,
Russell's blog[1] is down and there are not much information about
this any where else. Any one with more information about res_ais and
how it is used?
raj
[1]
http://www.russellbryant.net/blog/index.php/2008/06/10/asterisk-16-now-with-distributed-presence/
Really once I read credit card, I got to become interested to know whatis
exactly happenning.
I am looking to have the possibility to pay to the bank using the VoIP adaptor
or IP Telephony, by entering the credit card digits and the password and the
amound.
I do not know if u can help me in
Dear Alex,
Thanks for the reply..Can you please list some of these solutions that you
talked about on your reply?
Even I would like to ask if you had a bad experience with asterisk regarding
simultaneous calls limitation and If I'll send 1k calls to an asterisk
machine with the appropriate
Dear Helm,
Kindly confirm why you do not recommend the VMs solution and if you had bad
experience for it and what did you get?
Regards
On Tue, Feb 17, 2009 at 9:24 PM, Wilton Helm wh...@compuserve.com wrote:
You may be able to split up some of the servers into multiple VMs -- maybe
five
Hello list,
I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level.
If I use the following code:
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten = s,n,Queue(myQueue)
this works fine for the FIRST call
On 17 Feb 2009, at 19:20, David Gibbons wrote:
snip
We will be testing the ADT connection heavily this week. The modem
connections to my understanding are 2400 baud. Over G.711U and a T1 I
don't see why this wouldn't be as solid as a POTS line, but our
tests will
tell!
/snip
We do *fax*
On Wed, Feb 18, 2009 at 10:02:28AM +, Tim Panton wrote:
Our creditcard company's small print _insists_ on a direct analog
exchange line with no other devices in between.
Wow. You have a direct copper wire to their credit card processing
system? :-)
--
Tzafrir Cohen
Dear Sir,
Can someone help me please to find a free ebook talking about AGI scripting
through asterisk?
Regards
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To UNSUBSCRIBE or update options
How to convert SIP-T to SIP for Asterisk?
2009/2/17 Raj Jain rj2...@gmail.com
On Tue, Feb 17, 2009 at 1:06 PM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
Asterisk supports SIP-T?
Nope. Here is some old discussion on this topic:
On Wed, Feb 18, 2009 at 6:55 AM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
How to convert SIP-T to SIP for Asterisk?
You'll need to strip out ISUP MIME body in your SIP messaging with Asterisk.
--
Raj Jain
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http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
asterisk-support books section Asterisk: The Future of
Telephonyhttp://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
is greate.
David
2009/2/18 michel
On Tue, 17 Feb 2009, Gordon Henderson wrote:
It looks like something has changed in the HPET kernel code in 2.6.28
(maybe .27 too) that's stopped ztdummy.c compiling (in 1.2 and 1.4
versions of zapata) A kernel structure member has been renamed with some
crypic comments in the lkml about it.
Dear Sir,
the asterisk book from oreilly does not make full description to the AGI
scripting...I suggest please if someone advice to me a free PDF book just
dedicated for AGI and nothing else
Regards
On Wed, Feb 18, 2009 at 2:09 PM, David fire ddf...@gmail.com wrote:
Hi,
I would like to know if it is possible to detect which party initiates a
hangup - and if so, how this is done.
In my asterisk log, I see something like the following:
Feb 18 04:14:13 VERBOSE[17488] logger.c: -- Executing
Hangup(IAX2/ToHK1-16, ) in new stack
Feb 18 04:14:13
http://www.speechtechmag.com/Articles/News/News-Feature/Ditech-to-Delive
r-Voice-Based-Web-Interaction-during-Mobile-Calls--52606.aspx
thought this Ditech API might interest a few people.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357
Hi All,
Asterisk 1.4.12 CentOS 5
My ISP account includes nearly 500 minutes of VOIP calls per month but
the service is expensive for unbundled minutes. So I'm trying to find
a way to keep an accumulated total of calls made through that trunk so
that I can automatically switch to a lower-cost
Wilton Helm wh...@compuserve.com writes:
I'm not sure I see the merit in this. VMs seem to be regarded as a magic
bullet (i.e. free lunch). I don't know of any case where 5 VMs can
accomplish more work on one processor than simply letting the processor
manage it all
Modern machines have
You shoudl start with your bank. They can probably provide the equipment.
j
On Wed, 18 Feb 2009, bilal ghayyad wrote:
Really once I read credit card, I got to become interested to know whatis
exactly happenning.
I am looking to have the possibility to pay to the bank using the VoIP
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
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On Wed, Feb 18, 2009 at 3:42 AM, michel freiha mich...@gmail.com wrote:
Dear Alex,
Thanks for the reply..Can you please list some of these solutions that you
talked about on your reply?
Even I would like to ask if you had a bad
But clients can call to your IVR and do the payment from their home or the
mobile, correct?
If that is possible, how they pay $50.53? I mean, how they enter the 0.53$?
They use the * to express the (.)?
From the other side, they asking for analog telephone line or they need a
leased line
use the h exten.
when someone hangup dial go to exten h.
or put the option in the dial command to go to the next priority on hangup
but there is a problem if during the call they transfer it to other exten
you dont have the next priority.
David
2009/2/18 Geoff Lane ge...@gjctech.co.uk
Hi All,
Try www.cpan.org http://www.cpan.org/ -- modules -- agi. This will
help you from a PERL perspective. The AGI is also applicable for (at least)
PHP and C+.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Put this snippet in a macro and call the macro. That way the data lives
for the duration of the incoming call.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, February 18, 2009 3:05 AM
To:
Danny Nicholas da...@debsinc.com writes:
You could set up a hint for the extension and check the hint for inuse
before executing the Dial in your dialplan
Exten = 801,hint,SIP/100
Exten = XXX,1,System(/usr/sbin/asterisk -rx core show hints|/bin/grep
SIP/100|/bin/grep InUse
I think we have
Hi All,
I am using asterisk 1.4.19,
I have setup the dialplans to get the incoming call and that will be sent to
another context by local channel,
In another context i have setup the ring group, that portion is working
fine.
I have noticed that when i have set one of the extension in call
Lenz Emilitri lenz.lo...@gmail.com writes:
If I use the following code:
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten = s,n,Queue(myQueue)
this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated on subsequent
Nhadie wrote:
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
I've been happy with FlowRoute.
I have a number port pending with them that's taking forever (requested
in Oct), but I doubt it's their fault. In every other respect, I've been
happy
Hello,
I have a working system based on asterisk 1.4.23.1 and I want RTP going
end-to-end but not using canreinvite because it creates problems in my
configuration. I have tested directrtpsetup=yes and canreinvite=no but media
goes through asterisk. I know this feature is experimental, but so
Our creditcard company's small print _insists_ on a direct analog
exchange line
with no other devices in between.
Tim.
Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk
You can do it an interface using AGI to comunicate with equipment or verifone.
I did it
I think this is by design - each time the Dial() is performed, SIP headers
are reset.
l.
2009/2/18 Benny Amorsen benny+use...@amorsen.dk benny%2buse...@amorsen.dk
Lenz Emilitri lenz.lo...@gmail.com writes:
If I use the following code:
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
Interestiong - how would you do this? I thought macros on the queue command
were only for 1.6.
l.
2009/2/18 Danny Nicholas da...@debsinc.com
Put this snippet in a macro and call the macro. That way the data lives
for the duration of the incoming call.
--
On Feb 18, 2009, at 9:57 AM, Paul Chambers wrote:
Nhadie wrote:
Hi,
Anyone knows a DID provider that can do both outbound and inbound?
Regards
Nhadie
I've been happy with FlowRoute.
I have a number port pending with them that's taking forever
(requested
in Oct), but I doubt it's
What economic downturn?
I'm sick and tired of hearing this mantra.
I wish you the best of luck in maintaining your immunity.
Same here (in the UK).
As long as people need to make phone calls ...
Gordon
The economy (and indeed
I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300
and siemens hipath 4000. (2 channels to each switch)
with a TE210p card setup as T1 with em_w.
When the call is initiated to either switch the phone rings, when its
answered then nothing...
I hear no audio etc... After the
On Wednesday, February 18, 2009, David fire wrote:
use the h exten. when someone hangup dial go to exten h. or put the
option in the dial command to go to the next priority on hangup but
there is a problem if during the call they transfer it to other
exten you dont have the next priority.
Lets say your dialplan looks like this:
exten = s,1,Answer()
exten = s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten = s,n,Queue(myQueue)
exten = s,n,blah
exten = s,n,Hangup()
you would make a macro like this
[macro-siphead]
exten = s,n,SIPAddHeader(X-Unique-ID: ${ARG1})
exten =
On Wed, 2009-02-18 at 14:36 +0200, michel freiha wrote:
the asterisk book from oreilly does not make full description to the
AGI scripting...
You're right.. the O'Reilly book doesn't make a full and complete
description of AGI programming. (It was better than anything else
written at the time,
On Tue, 2009-02-17 at 12:24 -0700, Wilton Helm wrote:
I'm not sure I see the merit in this. VMs seem to be regarded as a
magic bullet (i.e. free lunch). I don't know of any case where 5 VMs
can accomplish more work on one processor than simply letting the
processor manage it all (except if
Michael wrote:
On Wed, 18 Feb 2009 13:37:57 John Todd wrote:
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
Sunday in Los Angeles, and the topic of my talk is Open Source in an
Economic Downturn. I've got lots of talking points for this talk,
but it would be
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or IP
Phone registered with Asterisk), and Asterisk communicate with the bank or
company credit card provider?
How the user will enter $50.25?
What about
Michael Graves wrote:
On Sun, 15 Feb 2009 10:23:50 +0800, Nhadie wrote:
Hi All,
If i buy 20 g729 and install to my asterisk, if 20 calls are already
engaged using g729. would the next call then revert to using the other
codec, in this case ulau and alaw?
Yes, if you set the codec
Thanks for the feedback. I did some research and it looks like you were calling
over international lines. It also appears that there was high than average
static on the line, which is not normal for my system. It's true that I threw
my recordings together quickly and the beep was supposed to be
bilal ghayyad wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank or
company credit card provider?
How the user will
Ideally the person needs to enter the credit card number, expiration date in
mmyy format (which is the format in which the expiration date is shown on
the card), and the ccv number. The amount would probably be calculated on
the basis of the outstanding amounts, or the products selected. Think
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of
Asterisk B (registered in Asterisk A as extension)
to incoming call across another trunk of Asterisk B to extension of Asterisk
C
What the dial plan should be?
Thanks
--
We never did too much talking anyway
So don't
On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk nt_aster...@yahoo.comwrote:
This module detects gender and approximate age range. I'm working on
getting it's accuracy to 80%+ on a consistent basis, after implementing
filters to remove background noise and other artifacts.
It's designed
On Wed, 18 Feb 2009, Richard Lyman wrote:
bilal ghayyad wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank
or company
Hi,
You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
It's a good place to start.
--
Regards,
Robert Broyles
Leonja Cerebro wrote:
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk
of Asterisk B (registered
Jeff LaCoursiere wrote:
On Wed, 18 Feb 2009, Richard Lyman wrote:
bilal ghayyad wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk
On Wed, Feb 18, 2009 at 1:28 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Mon, Feb 16, 2009 at 2:45 PM, Asterisk Asterisk
nt_aster...@yahoo.comwrote:
This module detects gender and approximate age range. I'm working on
getting it's accuracy to 80%+ on a consistent basis,
Hi List.
I'm having problems with Asterisk 1.6 + DAHDI 2.1.0.3
PlayBack does not ring, is still in command, and not later in the following
context.
Disabling the dahdi operates normally.
I'm using dahdi_dummy.
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On Wed, Feb 18, 2009 at 12:50, bilal ghayyad bilmar...@yahoo.com wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank or
Perhaps there isn't a downturn in a country with more sheep than people
That's a little harsh, New Zealand is one of those places that really appeals
as a decent place to live.
This could easily turn into a decade long period, with few having any real
answers.
I think that's somewhat
On Thu, 19 Feb 2009 08:33:56 Chris Bagnall wrote:
Perhaps there isn't a downturn in a country with more sheep than people
That's a little harsh, New Zealand is one of those places that really
appeals as a decent place to live.
It is. And when it isn't so hot (mid summer) Australia is also
I have five Asterisk servers running 1.2.14, and am planning to upgrade
to 1.4 this weekend. In preparation, to use the most efficient g729
codec, I am running the new benchg729 program. It works great on two
systems, but on the other three it says it cannot locate a valid g729
license. I have
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on
Sunday in Los Angeles, and the topic of my talk is Open Source in an
Economic Downturn. I've got lots of talking points for this talk,
but it would be interesting to hear some short anecdotes about how you
in the
I have problem of using call file to make auto outbound dial through FXO
channel. I put Channel: DAHDI/1/xx (xx is the
destination PSTN number to dial). For some reason asterisk did not dial
the number but the control came to the context that I defined in the call
file as if the
Adam Robins wrote:
I have five Asterisk servers running 1.2.14, and am planning to upgrade
to 1.4 this weekend. In preparation, to use the most efficient g729
codec, I am running the new benchg729 program. It works great on two
systems, but on the other three it says it cannot locate a valid
Asterisk-users,
Our two-part tutorial explaining how to use VoIP and Asterisk in
Amazon’s Elastic Compute Cloud (EC2) has garnered quite a bit of
attention. But due to the time required to complete the many steps
needed to get up and running, some of you have asked if it is possible
to
That did it. Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, February 18, 2009 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Feb 18, 2009, at 1:18 PM, Eric Chamberlain wrote:
Asterisk-users,
Our two-part tutorial explaining how to use VoIP and Asterisk in
Amazon’s Elastic Compute Cloud (EC2) has garnered quite a bit of
attention. But due to the time required to complete the many steps
needed to get up and
Hi!
I have some problems understanding the concept of REFER in Asterisk 1.4.23.
I have the following scenario:
Incoming SIP call (incoming leg) from a SIP trunk into Asterisk (handled
in context fromTrunk), forwarded to the SIP Client (outgoing leg).
Now, the SIP Client sends a REFER request
Ray Chen wrote:
I have problem of using call file to make auto outbound dial through FXO
channel. I put Channel: DAHDI/1/xx (xx is the destination
PSTN number to dial). For some reason asterisk did not dial the number but
the control came to the context that I defined in the
Steve,
Tried to test and got call could not be completed as dialed.
Were you able to connect? If not, please try again. Call volume has been
growing.
How about a moving stress variable that could be used as a lie detector
of sorts or
just to measure how certain parts of a script, or certain
Hi Justin,
How far is your work from being able to do speaker verification? Not
*identification* mind you, but being able to tell that a captured voice is
the same as another that is stored...
Cheers,
j
On Wed, 18 Feb 2009, Asterisk Asterisk wrote:
Steve,
Tried to test and got call
I hadn't even thought of that, but that's a great idea. I wrote some code that
does speech recognition based on generated tokens and no learning needed. We
could certainly apply the gender detection and that sr to a project like this.
I would store only the token in my current model, but we
Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk Asterisk wrote:
Steve,
Tried to test and got call could not be completed as dialed.
Were you able to connect? If not, please try again. Call volume
On Wed, 18 Feb 2009, Danny Nicholas wrote:
The AGI is also applicable for (at least) PHP and C+.
An AGI (or more accurately, an executable conforming to the AGI
specification) can be written in any language -- Fortran, assembly, shell
script, BLISS (if you wanted to fastagi over to a VAX
On Wed, 18 Feb 2009, michel freiha wrote:
I suggest please if someone advice to me a free PDF book just dedicated
for AGI and nothing else
It takes a rare individual to put the effort required to write a book and
then distribute it for free.
I would like to write it, but my kids have grown
I think PHP is the best language to write AGIs, there is a library available
(PHPAGI) and it is easier to work with as compared to the complexity of the
C language. That book written by Nir Simionovich (Asterisk AGI 1.4 and 1.6
Programming) clearly describe the pros and cons of writing AGI
There's a cheaper pdf version of his book as well. You can get more info
at:
http://www.packtpub.com/asterisk-gateway-interface-programming/book
On Wed, Feb 18, 2009 at 7:59 PM, Emmanuel Bruno tipas...@gmail.com wrote:
I think PHP is the best language to write AGIs, there is a library
Just for the records,
Those of you who likes PHP simplicity (I'm one of them :) ) , but would
like C small footprint, and almost *incomparable speed*; there's a PHP
compiler that takes PHP code to C, and produces an optimized executable.
You can see it here:
On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote:
On Wed, 18 Feb 2009, michel freiha wrote:
I suggest please if someone advice to me a free PDF book just dedicated
for AGI and nothing else
It takes a rare individual to put the effort required to write a book and
then distribute it for
Michael wrote:
On Thu, 19 Feb 2009 13:35:25 Steve Edwards wrote:
On Wed, 18 Feb 2009, michel freiha wrote:
I suggest please if someone advice to me a free PDF book just dedicated
for AGI and nothing else
It takes a rare individual to put the effort required to write a book and
then
Kevin P. Fleming wrote:
Muiz Motani wrote:
I understand that the Asterisk SLA implementation is somewhat different
from most key systems and PBX systems. I also understand that in
Asterisk, one does not put an SLA line on hold since it is just a MeetMe
conference. However, is there any
I would like to write it, but my kids have grown accustomed to eating :)
This is everything that is wrong with Open Source - no body wants to pay
for anything
Hi my name is nobody. I like to pay for many (FOSS) things. Not as
much as I'd like but when the right Powerball ticket
Hi All,
I have configured the phpagi application for counting the duration of call,
The call is originated from the script and after hangup the call the
duration and status will be stored.
This functionality and php script is working fine with deadagi application
with asterisk 1.4.
I have a
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier a écrit :
just for curiosity, is AOC-E messages sending included in telco basic
subscription or is an option needed for that ?
cheers
It's included on PRI (even partial) subscriptions, optional on BRI.
Thanks,
- --
Jean-Denis Girard
Hi,
I've been asked sometimes to tailor call history features embeded in SIP
hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific criteria.
1. Have you identified a phone offering the possibility to display as Call
History, an
2009/2/19 Jean-Denis Girard jd.gir...@sysnux.pf
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier a écrit :
just for curiosity, is AOC-E messages sending included in telco basic
subscription or is an option needed for that ?
cheers
It's included on PRI (even partial)
18 feb 2009 kl. 13.07 skrev Raj Jain:
On Wed, Feb 18, 2009 at 6:55 AM, Daviramos Roussenq Fortunato
daviramo...@gmail.com wrote:
How to convert SIP-T to SIP for Asterisk?
You'll need to strip out ISUP MIME body in your SIP messaging with
Asterisk.
I don't think you need to strip it out
19 feb 2009 kl. 00.08 skrev Klaus Darilion:
Hi!
I have some problems understanding the concept of REFER in Asterisk
1.4.23.
I have the following scenario:
Incoming SIP call (incoming leg) from a SIP trunk into Asterisk
(handled
in context fromTrunk), forwarded to the SIP Client
19 feb 2009 kl. 07.47 skrev Olivier:
Hi,
I've been asked sometimes to tailor call history features embeded in
SIP hardphones.
For example, a cutomer wanted internal call to be taken out.
Another wanted calls to sorted according specific criteria.
1. Have you identified a phone offering
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