On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote:
Thanks William and John, I'll look again for that download. Comments
below...
--On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett
[EMAIL PROTECTED] wrote:
FWIW? I bought that voice and I find it amusing, but not ready for
prime time.
On 7/15/05, Mark Edwards [EMAIL PROTECTED] wrote:
Yes!
is Vonage SIP or IAX Terminated? I am experiencing the exact same issue and
I have logged a bug
Does Vonage work with Asterisk? How much is this type of plan from Vonage?
Thanks,
Michael
Hi,
What would you say that the best compression format is for voice
recordings on Asterisk? The tradeoff being the file's size. I like
GSM because of the small files size but the quality isn't great. What
are people finding as a good setting for GSM?
Thanks,
Michael
Thanks!
What settings are you using for GSM , bit rate, etc?
Thanks,
Michael
On 5/18/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On May 18, 2005 10:53 am, Michael Stearne wrote:
What would you say that the best compression format is for voice
recordings on Asterisk? The tradeoff being
PROTECTED] wrote:
On May 18, 2005 02:59 pm, Michael Stearne wrote:
What settings are you using for GSM , bit rate, etc?
I'm not setting anything, just allow=gsm.
-A.
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I am developing an Asterisk based IVR system using IAX channels in and
out. Is there anyway to develop and test Asterisk on a local machine
using say a soft phone? I just want to be able to develop the app
while on the road (with no Internet access). Is this possible?
Thanks,
Michael
:
[outgoing]
exten = s,1,Dial(IAX2/Fom2QqL88D:[EMAIL PROTECTED]/${EXTEN})
Again I am very new. :-)
Thanks,
Michael
On 5/20/05, Michael Stearne [EMAIL PROTECTED] wrote:
I am developing an Asterisk based IVR system using IAX channels in and
out. Is there anyway to develop and test Asterisk
Is there anyway to NOT allow the incoming caller to leave a voicemail
message for a certain mailbox? I would like the caller to hear the
message and then have the option to press 1(for example) to call the
user (make an outgoing call), but not to be able to leave the message.
Even if after the
-users-
|[EMAIL PROTECTED] On Behalf Of Michael Stearne
|Sent: Friday, May 20, 2005 11:42 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion; Jim Ginn
|Subject: [Asterisk-Users] Voicemail With No Messages?
|
|Is there anyway to NOT allow the incoming caller to leave a voicemail
Is there any way to have the user confirm the extension they are
looking to go to before transfering?
i.e.
You pressed 5 4 3 3 2. Is this correct?
1 - GoTo extensionPressed
2 - Enter extension again
Thanks!
Michael
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Stearne
Sent: Saturday, May 21, 2005 11:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Confirmation Of Extension Before Transfer?
Is there any way to have the user confirm the extension
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
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On 5/24/05, Roman Volf [EMAIL PROTECTED] wrote:
Michael Stearne wrote:
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
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.
All of this can be found with a simple web search.
On 24-May-05, at 11:12 AM, Michael Stearne wrote:
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
On 5/24/05, Matthew Boehm [EMAIL PROTECTED] wrote:
Michael Stearne wrote:
Does anyone have the MySQL add-on as a binary for OS X? Or am I
getting it wrong and MySQL is installed by default?
Thanks.
Michael
Hey Michael,
Seems some other people don't read posts before posting replys
Hi,
I am a newbie and just discovered AGI (after learning a lot about
extensions.conf's language). Before putting in a lot of time on
AGI/Perl/PHP I would like to know if its possible to do most of the
functionality performed in extensions.conf through AGI. Can AGI be
used as a replacement for
On 5/25/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
On Wed, 2005-05-25 at 00:14 -0400, Michael Stearne wrote:
Hi,
I am a newbie and just discovered AGI (after learning a lot about
extensions.conf's language). Before putting in a lot of time on
AGI/Perl/PHP I would
On 5/26/05, Jon Farmer [EMAIL PROTECTED] wrote:
Now the script loops forever while the user is connected and exits if
the user hangs up.
Thanks to everyone who helped me out, much appreciated.
Jon,
What version of PHPAGI are you using? I am starting a PHPAGI app and
want to know
released yet. I spoke to the developer and he
suggested 2.0.
Thanks,
Michael
Ben
On 5/26/05, Michael Stearne [EMAIL PROTECTED] wrote:
On 5/26/05, Jon Farmer [EMAIL PROTECTED] wrote:
Now the script loops forever while the user is connected and exits if
the user hangs up.
Thanks
On 5/26/05, Matthew Asham [EMAIL PROTECTED] wrote:
On Thu, 2005-26-05 at 17:59 -0400, Michael Stearne wrote:
On 5/26/05, Benjamin West [EMAIL PROTECTED] wrote:
Michael,
The version, in the context of Jon's problem, was irrelevant. Jon's
problem was due to a small bug in his code
I am using Asterisk 1.0.7. When running the console in asterisk -vvc
mode I get warnings about:
No log handling enabled - turning on stderr logging
Cannot find module (NET-SNMP-EXTEND-MIB): At line 0 in (none)
Is there any way to correct this warning. What am I missing that I
need to install?
In our IVR we have a user enter a 6 digit number and information if
returned. Our problem is that no all of the digits that the user
presses are being recevied correctly. Its not as if the first digits
are being cut off or the last, its just some digits aren't coming
through.
Our setup is that
On 6/1/05, Jeff Heath [EMAIL PROTECTED] wrote:
I read on the Wiki that Asterisk Realtime requires CVS HEAD, but I've
also discovered that not everything on the Wiki is 100% accurate (that's
not a knock, but with a program that is changing as fast as Asterisk,
it's impossible for the
On 6/1/05, PA [EMAIL PROTECTED] wrote:
I've gotten my CDR working the way I like, but I am looking to customize it a
bit. I have set up an IVR menu, which works great. I would like to be able
to capture the prompted DTMF digits pressed by callers, to my CDR database
but I don't see any
On 6/1/05, Luis Diaz [EMAIL PROTECTED] wrote:
how can i do to display a message to every wrong number ???
I do something like:
$expandedNumbers=;
$result = $agi-get_data('beep', 4000, 6);
$numbersPressed = $result['result'];
for($i=0;$istrlen($numbersPressed);$i++){
On 6/1/05, Russell Bryant [EMAIL PROTECTED] wrote:
I am on IRC as drumkilla and also available by email if anyone has any
questions or comments.
Please test and report any issues on the Asterisk issue tracker, even if
it is just a note saying that you have no problems at all! I will
On 6/2/05, Mike M [EMAIL PROTECTED] wrote:
=
NOTE: Using older version of expression parser. To use the newer
version,
NOTE: upgrade to flex 2.5.31 or higher, which can be found at
NOTE:
I am trying to configure RealTime Voicemail with MySQL. I downloaded
compiled and installed the CVS HEAD for asterisk, and for
asterisk-addons. MySQL seems to be loading correctly (the cdr table
is recording incoming calls). But the RealTime Voicemail doesn't seem
to be checking the database
My fault! I was pointing to the wrong database in the res_mysql.conf file!
Setting debug mode in /etc/asterisk/logger.conf is very helpful.
Michael
On 6/2/05, Michael Stearne [EMAIL PROTECTED] wrote:
I am trying to configure RealTime Voicemail with MySQL. I downloaded
compiled
On 6/3/05, Asterisk User [EMAIL PROTECTED] wrote:
Hi experts,
I wish someone would kindly give me a hand on a problem on Asterisk
Realtime.
May I know how to enable the debug messages for the Asterisk SIP Registrar
query the SIP user data in the created MySQL table. I found that
On 6/6/05, Colin Anderson [EMAIL PROTECTED] wrote:
I'm just wondering if anyone in the community has considered what if and
what would be a meaningful response, either technologically, legally, or
socially. Encryption comes to mind. Also, Dundi's RFC perhaps addresses some
of these issues by
I am trying to use swift in PHP/AGI.
function swift($text, $escape_digits='', $frequency=8000, $voice=NULL,
$fnameIn='')
During swift speaking some text I want the caller to be able to press
1, 2 or 3 to do thing 1, thing 2 or thing 3.
How are these digit defines and then caught?
Thanks,
On 6/10/05, Clarke Kawakami [EMAIL PROTECTED] wrote:
Michael...
I don't believe that PHPAGI supports this currently. What you are looking
for is a combination of 2 functions: get_data() and swift().
That's what I was beginning to think but kept getting thrown off by
the escape digits
We are developing an IVR application and when I am testing locally on
my machine using a softphone (iaxcomm) the digits I press for GET DATA
work every time. I am testing with a local extension that goes right
into my routine. However when I try to call in to the system using an
analog or cell
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I am wondering if anyone else is experiencing similar issues. I
believe the problem lies with VoicePulse because we are using them for
IAX connections. I don't believe its a bandwidth problem on my
network (cable) because I have tried
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
That's entirely possible. Had something similar with livevoip.com (with
the answered iax call issue).
You should be able to determine whether its a voicepulse issue by either
doing a iax debug (look for the dtmf digits), or, using ethereal
are using sip, then in
sip.conf
regards,
Umair bari
Michael Stearne wrote:
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
That's entirely possible. Had something similar with livevoip.com (with
the answered iax call issue).
You should be able to determine whether its
If a user has created an unvailable message in Comedian mail is there
anyway to delete that message? I know you can record a new message,
but I would like to delete the file as if the user never recorded one.
Thanks,
Michael
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Asterisk-Users mailing
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I don't believe relaxdtmf is a valid parameter for iax.conf; just
sip.conf.
(per the most recent sample configs)
I didn't find it either. I put it in the config anyway but it didn't
seem to make a difference. I also tried changing
I just signed up and configured a SIP connection from BroadVoice. It
works great. This issue I have is that it seems after a couple calls
(or a certain amount of time) Asterisk doesn't seem to be receiving
these calls anymore. It seems as if BroadVoice is not redirecting the
call to my
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
If you are not seeing any of those, then voicepulse is sending the dtmf
via inband audio tones. The accuracy of inband audio tones will be less
then if the dtmf digits are sent within iax packets (Type: dtmf). If they
are arriving via
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I just signed up and configured a SIP connection from BroadVoice. It
works great. This issue I have is that it seems after a couple calls
(or a certain amount of time) Asterisk doesn't seem to be receiving
these calls anymore. It seems
On 6/11/05, Rich Adamson [EMAIL PROTECTED] wrote:
I just signed up and configured a SIP connection from BroadVoice. It
works great. This issue I have is that it seems after a couple calls
(or a certain amount of time) Asterisk doesn't seem to be receiving
these calls anymore. It seems
On 6/15/05, Darren Ellis [EMAIL PROTECTED] wrote:
Hi,
If there's anyone out there who has successfully compiled * 1.0.7 on
10.4.1, could you contact me off-list? I've tried the astmasters
mailing list, but it continually rejects my messages.
In order to get the CVS-HEAD to compile I had
On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote:
I'm planning an Asterisk Voicemail system of around 3000 users spread
across several sites, each site connected by a fast network to a central
site. We're considering 2 models:
- Central Voicemail with VoIP calls from remote sites
On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote:
Several people have responded with architecture suggestions. While these
are welcome, I'm happy with the architecture options planned, having
done many large voicemail implementations on products other than Asterisk.
What I had hoped
On 6/16/05, Bill McLaughlin [EMAIL PROTECTED] wrote:
Vonage uses Asterisk, and they have a lot more than 3000 customers.
That should help your argument!
Michael
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On 6/26/05, harry gaillac [EMAIL PROTECTED] wrote:
Hello,
I read
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail
however when i run show voicemail users app voicemail
return users in voicemail.conf
Why?
You should enable debugging in the console (logger.conf)
We have a basic application that runs a SIP channel to pick up a call
and process it. We are using Broadvoice and it's been working great.
We recently rebooted the machine and now when a call comes in Asterisk
picks up the call but does not process it. Asterisk seems to send the
call back to
Thanks!
In my dialplan there was no rule for 6092991xxx
Michael
On 9/18/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sun, 18 Sep 2005, Michael Stearne wrote:
Looking for 6092991xxx in from-broadvoice
Reliably Transmitting (no NAT) to 147.135.20.128:5060:
SIP/2.0 404 Not Found
We are trying to determine how to build out an IVR system we are
working on. The system needs to be able to handle probably at most
5-10 concurrent calls at peak times. Other times we are just looking
for a reliable service. For incoming calls we've been using
BroadVoice VOIP and before that
Does Asterisk work with Avaya? If so, is there any documentation on it?
Thanks,
Michael
On 10/9/05, Andy Vega [EMAIL PROTECTED] wrote:
Does anybody know if Avaya has a test SIP firmware available for 4620 and
4640 IP phones? The 46xx SIP image from their website is a combo download
with
Chris... thanks for the great reply
On 10/5/05, Chris Shaw [EMAIL PROTECTED] wrote:
Michael,
Doing an All-Network setup is completely doable but there are many factors
to consider.
First of all, I didn't see any mention of how many connections it takes
before Asterisk starts having
On 10/9/05, Andy Vega [EMAIL PROTECTED] wrote:
On 10/9/05, Michael Stearne [EMAIL PROTECTED] wrote:
Does Asterisk work with Avaya? If so, is there any documentation on it?
Thanks,
Michael
It does:
http://www.voip-info.org/tiki-index.php?page=Avaya+4602+configuration
Unfortunately
On 10/12/05, Mir [EMAIL PROTECTED] wrote:
Hello
We have discovered a problem with DTMF on Asterisk.
We have a setup with a T1 from PSTN going into an Asterisk box, and
then out again on T1 and into a normal PBX (EADS)
We use it to record all calls going to/from the PBX.
The problem is
I am trying to determine the number that was called in via an IAX2 channel.
When using debug:
IAX2 Debugging Enabled
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00013ms SCall: 00814 DCall: 0 [66.234.228.170:4569]
VERSION : 2
CALLED
From the console if there a way (in debugging or someting) to get a
list of currently defined Global/Local variables like CALLERID, etc?
Thanks,
Michael
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,
Michael
On 12/23/05, Michael Stearne [EMAIL PROTECTED] wrote:
From the console if there a way (in debugging or someting) to get a
list of currently defined Global/Local variables like CALLERID, etc?
Thanks,
Michael
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On 12/23/05, Michael Stearne [EMAIL PROTECTED] wrote:
On 12/23/05, C F [EMAIL PROTECTED] wrote:
show functions will give you lots of info and so will help.
/usr/src/asterisk/docs/README.variables
Thanks. Where can I download the docs? I didn't seem to get that
with the source I
On 12/23/05, Johann [EMAIL PROTECTED] wrote:
Michael Stearne wrote:
On 12/23/05, Michael Stearne [EMAIL PROTECTED] wrote:
On 12/23/05, C F [EMAIL PROTECTED] wrote:
show functions will give you lots of info and so will help.
/usr/src/asterisk/docs/README.variables
Thanks. Where can I
On 1/3/06, Bogdan Moldovan [EMAIL PROTECTED] wrote:
IMHO use FC4.
Also after the install of the OS and all the required packages do a 'yum
update'.
I am using FC3 right now with 1.0.9 and I am having a problem updating
to 1.2.1. I am trying to avoid upgrading to FC4 and I'll try a yum
update
On 1/3/06, Technical Support [EMAIL PROTECTED] wrote:
We do a lot of installs on Fedora (slowly becoming our favorite). Initially
clients asked for FC because of compatibility with Red Hat, great package
management, etc. With FC4, you get a great set of packages, and not a lot
of add-ons
I am having trouble with FC3.
After doing a yum update (of 1264 packages) I still cannont compile
1.2.1 from source:
make[1]: `libedit.a' is up to date.
make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast'
make[1]: `libdb1.a'
Thanks! I'll try that.
On 1/3/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
Have you removing the asterisk include directory before trying version
1.2? I think it might be /usr/include/asterisk/ in many cases.
Michael Stearne wrote:
I am having trouble with FC3.
After
On 1/3/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
Have you removing the asterisk include directory before trying version
1.2? I think it might be /usr/include/asterisk/ in many cases.
Thanks. Looks like this and make clean worked.
Michael
Michael Stearne wrote:
I am
On 1/4/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jan 03, 2006 at 06:43:16PM -0500, Michael Stearne wrote:
and when I try to update from binary:
[EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm
warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature
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