l
allow=ulaw
context=incoming
reinvite=no
canreinvite=no
nat=no
directrtpsetup=yes
rfc2833compensate=yes
rtpkeepalive=60
Thanks in advance!
- Justin Tunney
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I actually meant to take that out before copy and pasting. It was
just a zany option I tried to see if things would get better but they
didn't.
I will post a bug later this week about this.
On 3/30/07, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
Justin Tunney wrote:
> rfc2833
You should post this on the bugtracker (bugs.digium.com) along with a back
trace. See asterisk-sources/doc/README.backtrace for info on how to do a
backtrace.
Justin Tunney
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A
How are you gentlemen.
All your VoIP are belong to us.
Signed,
China
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I have just written a patch for Asterisk 1.2.0-rc1 that allows you to
install a voice changer on a channel.
http://www.lobstertech.com/voicechanger/
If you are a developer, please feel free to help add features and
clean up the patch so that we can hopefully get it in CVS some day.
- Justin
A little off-topic but I actually got 6 TDM400P cards in one system to
all be on their own interrupts using this "P4 Photon V1.01P" 865chip
board. The last card shared interrupts with some other stuff that
wasn't interrupt intensive. The system was stable ringing 12 phones
24/7 in 100 degree heat
How about you stop pulling your hair out and let me send you one of the
56k modems I have sitting on my desk. heh
On Fri, 11 Nov 2005 13:00:15 -0500, Branko Samardzic
<[EMAIL PROTECTED]> wrote:
Hi,
I am wondering if it is possible to tweak IAX2 protocol to packetize
audio
data
more ef
i386 GNU/Linux
with Asterisk 1.2.13.
Also, I am not using a zaptel timer. Could this possibly be causing
problems with DTMF??
Thanks!
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Migrating to 1.4 is not an option. I don't know what that is, but I
doubt my voip provider supports it.
On 11/8/06, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
Have you tried 1.4 with vldtmf?
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ereal that will give me debug
information on RFC2833 in the RTP stream?
Thanks,
Justin Tunney
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iew.php?id=5970
Direct link to patch for the lazy:
http://bugs.digium.com/file_download.php?file_id=11337&type=bug
On 11/8/06, Justin Tunney <[EMAIL PROTECTED]> wrote:
Asterisk People,
I'm currently using Asterisk and with a SIP voip provider and I'm
having problems where
o do so. The problem you are having right
now is one of many reasons why I feel this way.
I believe you can do this by editing your yum config and throwing in:
exclude=kernel-*
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On Tue, 21 Mar 2006 12:56:13 -0500, Fernando Lujan
<[EMAIL PROTECTED]> wrote:
Is it possible to store voicemail recorded messages using odbc?
Fernando Lujan
see asterisk-sources/doc/README-odbcstorage
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. This doesn't look like standard behavior to me. Is this some
sort of a master process?
yes, the script "safe_asterisk" will respawn asterisk.
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Asteri
l /tftpboot/
total 3208
-rw-r--r-- 1 root root350016 Jan 20 12:55 randomfirmware.bin
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On Wed, 29 Mar 2006 14:19:41 -0500, Matt <[EMAIL PROTECTED]> wrote:
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
Where did you hear this? Can you give a link? Looks like I'm going to
jail, tee hee.
On Wed, 29 Mar 2006 14:54:19 -0500, Matt <[EMAIL PROTECTED]> wrote:
IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone
service and wants his CID to read 'Jane Smith', who is Mary's sister,
so that Mary will answer when Joe calls.
I was always under the impression that the telcos
terisk
bridge the calls?
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Justin
into MusicOnHold,
which I use for testing new MOH music. I've noticed a similar thing with
this, that sometimes it finished playing one track, and just doesn't
start the next.
Alex
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Justin Tunney
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I highly recommed against using hyperthreading. It always seems to cause
intermittent kernel panics for me when I forget to turn it off.
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ing analog callers
- Analog callers can not hear each other in conference
This seems to happen with the 4-port and 24-port TDM cards sold by Digium.
Has anyone else experienced similar problems?
Thanks!
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e CPU to spare and mpg123
installed via app_mp3. You must also have Festival installed on your system
so the Jukebox can generate text to speech.
- Justin Tunney
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First, you want to set verbose to AT LEAST 666. You may also want to
turn on debug messages in logger.conf
Most of my DTMF problems stem from a mismatch between the type of VoIP
DTMF asterisk is expecting and for what the phone is configured.
On 8/16/06, Miguel Ruiz Velasco <[EMAIL PROTECTED]>
On Thursday 17 August 2006 17:12, Douglas Garstang wrote:
> Ok, maybe I'm having a brain fart, or maybe I've never gotten quite this
> far, but, if you call a fast AGI script, how do you RETURN data from the
> fast AGI back to the dialplan???
You could set some channel variables.
_
On 8/19/06, Dinesh Nair <[EMAIL PROTECTED]> wrote:
why would you want to run asterisk on the phone ? ideally, it should be
running a softphone and connecting back over WiFi or 3G (HSPDA ??) to an
asterisk installation.
You're thinking too logically! Open source programmers just wanna have fun
On 8/20/06, Lito Lampitoc <[EMAIL PROTECTED]> wrote:
I am connecting 80 locals with 16 PSTN lines. Which means, i need 4 digium
cards with 4 FXOs per card. All 80 locals will be connected to ATA devices.
It is not a good idea to put 4 Digium cards in one machine. Perhaps
you could try the TDM2
On 8/20/06, Steve Gladden <[EMAIL PROTECTED]> wrote:
Is it a bug or is it me?
The voicemail system in asterisk is very buggy. Can you show us the
text config file thingy that's associated with the voicemail message?
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Did you put:
include => parkedcalls
in your dialplan?
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I can fix it so others don't have the same problem. I
usually respond pretty quick to email (in the 5 minutes to one day
range)
Plans for 0.6:
- Change voice pitch via manager api and command line
- Open to suggestions
Have fun!
- Justin Tunney
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On 8/26/06, Peder @ NetworkOblivion <[EMAIL PROTECTED]> wrote:
Who says * isn't stable enough for prime time? At least it is on 1.0.3.
What kind of abuse does that box take?
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asterisk-use
To anyone having problems installing SoundTouch or libsoundtouch4c,
I've improved the build system for libsoundtouch4c and updated the
install instructions. Please let me know if you continue to have
problems.
http://www.lobstertech.com/code/voicechanger/
- Justin
__
Can you give us some more info? Like agi debug output?
On 8/27/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hi,
I'm having problems with calling the ${CDR(billsec)} & ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
users/pee
Stop trying to con lonely nerds in to answering you questions with
subjects like that Steve!
Anyway, check the bug tracker, I think someone posted on this list
about a week ago with the exact same problem.
On 8/27/06, Steve Gladden <[EMAIL PROTECTED]> wrote:
Is it a bug or is it me?
For the lo
Lobster Technologies has just anounced the release of the most
annoying open source IVR application ever devised by lobsters called
"PhoneParrot". PhoneParrot is an app that uses silence detection to
repeat everything a person says in to the phone.
http://www.lobstertech.com/code/phoneparrot/
F
On 9/4/06, Elpidio Ramos <[EMAIL PROTECTED]> wrote:
When attempting to do the same from outside my network (from my dsl
connection from home), I get to hear the asterisk auto attendant but not
able to send any sound from my laptop.
ARE YOU SURE IT ISN'T A DTMF
PROBLEM!!?
Check /etc/asterisk/modules.conf and see if there is a line trying to load it.
On 9/18/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
[EMAIL PROTECTED] asterisk]# /etc/rc.d/init.d/asterisk start
Starting Asterisk PBX: FATAL: Module ixj not found.
__
, please don't post details of this yet, keep it in the
mailing list for now. I will make a press release when I feel it
works.
- Justin Tunney
On 9/27/06, Adi Simon <[EMAIL PROTECTED]> wrote:
Hi,
Did anyone actually manage setting up a single SER with multiple Asterisk
boxes?
I
The last Asterisk system I developed had a whopping 0 dialplan
entries. It generated calls via the manager interface and handled
them through agi.
Does that make me a completely emasculated Asterisk hacker?
On 10/10/06, Steve Murphy <[EMAIL PROTECTED]> wrote:
Hello!
In my relentless quest for
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