Hello,
Does anyone know if a GUI for Asterisk exists ?
Regards,
Abdul Hakeem
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hank
Sent: 10 March 2004 01:43
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] From 0 to PBX in 2 hours
what is your easy
i am running asterisk, CVS -02/24/04 -13.55.19 (version)
i am have a voicetronix openswitch12 card.
i have installed the driver withmodprobe it loads fine.
But when i run asterisk i get a seg fault, this
seg fault occurs at different parts of thye running asterisk . for example
on a first run
Hello..
I just wrote a small application and i'm having
problems sending voice frames to channels..
Here is a little debug:
Mar 10 11:22:52 DEBUG[311316]: app_hotline1.c:836
hotline_exec: Writing frame on OH323/R20723 from OH323/L9507 ,Frame
type:2,Len:320,Mallocd:0
Mar 10 11:22:52
Hello,
For our company we will use Asterisk to receive our faxes. We would like
to manage this faxes. To give some information to a fax. Here fore I
wrote a little application.
To manage the incoming faxes I wrote a script which checks the directory
where the faxes are stored. If there is a
I've got a VP206 here. Want it? :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Abdul Hakeem
Sent: Wednesday, March 10, 2004 3:54 AM
To: Asterisk Users
Subject: RE: [Asterisk-Users] From 0 to PBX in 2 hours
Importance: High
Hello,
Does
Hi Johan,
-Original Message-
For our company we will use Asterisk to receive our faxes. We
would like
to manage this faxes. To give some information to a fax. Here fore I
wrote a little application.
I'd love to get to work on this, but how do you handle receiving the faxes
with
All of the numbers he's showing are apparently adding inbound and outbound
traffic together, giving results that are approximately double what is
actually seen on the wire. If he is working in a half-duplex ethernet
environment, those numbers have some meaning; if full-duplex, then cut them
in
Hello all,
asterisk-oh323 has been updated. The new version 0.5.10 fixes
the incorrect answering of H.323 channels (thanks to the people
of the list who helped to trace the problem). Also, I have added
support for Gnomemeeting text messages (just for fun).
Additionally, the new version contains
i'm looking for a G.729 codec that will work with an IAX
client. anyone have any ideas here?
alex, enjoyed the article -thecodecs are
explainedathttp://www.cisco.com/warp/public/788/voip/codec_complexity.html
thx,
kaydon
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I posted the results of my real world analysis of codec bandwidth usage on this list a
couple of weeks back. Here's the table I put together and an example of calculating
bandwidth over ADSL.
G.711 over Ethernet = 95 Kbps per channel
G.711 over IP/PPP = 86 Kbps per channel
I have applied the patch and restarted Asterisk.
But it still only requires a single # to transfer.
Did I possibly miss something?
This is just to show that it was applied.
[EMAIL PROTECTED] asterisk]# pwd
/usr/src/asterisk
[EMAIL PROTECTED] asterisk]# patch -p0 ../old_asterisk/doublehash.patch
the phone works for the wlan600 its a great phone poor battery but even
palms with wifi use ALOT more battery when wifi is on and considering
this phone has the wifi ON all the time the 23 stand by hours and 3 hr
talk is ok
it registers with asterisk just fine, try get it from pulver
Miguel
On
I had fairly good experince at first in the lab.
But when I configured the phone to use 104-bit WEP-key,
like most of the production networks,
the quality degraded a lot. You can still talk but the quality is bad (
choppy ).
With 40bit key or without WEP, the quality was fine.
I tested this a
Hi,
Anyone knows what needs to be changed in sipura dial plan:
(*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.)
In order to dial *.
Ta
SJ
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On 10/03/04 08:06, Augustine Olaifa wrote:
cntext=internal
[...]
sgnalling=fxo_ls
[...]
signallng=fxs_ls
[...]
i do not know what i am doing wrong?
I know English probably isn't your first language, but try learning how
to spell. ;)
Alastair
___
Nobody had some expirience with it?
Begin forwarded message:
From: Osvaldo Mundim [EMAIL PROTECTED]
Date: March 9, 2004 6:28:04 PM GMT-03:00
To: [EMAIL PROTECTED]
Subject: got (-2) from Queue
Hi all,
When I call Queue application from AGI, I always got (-2) as returned
value. Seeing the show
Hi,
Yes, and many thanks for the offer.
Can you email or arrange for FTP ?
Cheers,
Abdul Hakeem
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Marlowe
Sent: 10 March 2004 11:31
To: Asterisk Users
Subject: RE: [Asterisk-Users] From 0 to PBX in 2
You must make sure that your application generates voice frames
with the proper rate. If you are too fast, then the outgoing
channel will experience overruns (this seems to happen from your
debug messages).
Michael.
Alexandru Coseru wrote:
Hello..
I just wrote a small application and i'm having
John Congdon wrote:
I have applied the patch and restarted Asterisk.
But it still only requires a single # to transfer.
Did I possibly miss something?
This is just to show that it was applied.
[EMAIL PROTECTED] asterisk]# pwd
/usr/src/asterisk
[EMAIL PROTECTED] asterisk]# patch -p0
You are informing for Asterisk that your E1 channels will use FSX Loop Start
signalling and the boards are expecting ISDN PRI signalling. Try to put a
PRI_CPE signalling in your configuration. Something like that:
.
.
.
group=2
switchtype=euroisdn
signalling=PRI_CPE
pridialplan=international
On Wed, 10 Mar 2004, Fran Boon wrote:
Patch failed - this is what this output is showing.
As Matt said the patch needs modifying to patch cleanly against the
current version of the code...
You didn't read his mail properly.
Steve
___
I'm trying to set up my first phone, Windows Messenger running on an
attached PC.
It's talking to the server, but the server never responds. Instead, it
gives this error:
Mar 10 06:26:20 WARNING[163851]: chan_sip.c:455 __sip_xmit: sip_xmit of
0xbe5fa5e0 (len 490) to 192.168.1.99 returned -1:
Hi list,
Faxes come in over an E1 line (on an TE410P) here and then are sent to
an analog fax machine attached to a T1 (also on the TE410P)
channelbank (CAC1).
Problem is that almost all faxes we send out and receive are mangled...
either only halve pages or very stretched text etc.
Setup in
But why it is hanging up the channel ?
It should not send the overflow data..
Alex
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 3:31 PM
Subject: Re: [Asterisk-Users] Ast_smoother_feed problems
You must make sure
Michiel,
Are you using WinFax? or one of the Products Based on Winfax? I've seen this on all
of our WinFax Stations, but none of our standalone Fax machines.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of michiel betel
Sent: Wednesday, March 10,
My mistake...
I was closing the channel..
Sorry for that
Regards
Alex
- Original Message -
From: Alexandru Coseru [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 4:45 PM
Subject: Re: [Asterisk-Users] Ast_smoother_feed problems
But why it is hanging up the
Hi!
When I try to call from a SIP phone to a PBX phone I get this error:
chan_oh323.c [1004] Couldn`t call 483377839
and if I get the messages from SIP debug, I have a 403 message. The
configuration of my system is:
SIP Phone ASterisk Gatekeeper - Gateway - PBX - Phone
Hello,
I'm still doing some tests with Asterisk before reaching a production state.
To do some VoIP-PSTN tests I'd like to know how to configure Asterisk
to use an ISDN card such as Eicon Diva Server BRI 2M.
Any hints are much appreciated.
Martin
Hi,
I ordered the * Developers Kit which I though would be able to handle future expansion
by the phrase Single-Port (upgradeable to 4) TDM400P but I've never seen the modules
for sale anywhere.
Can someone enlighten me please as to where they can be had and for what price?
tia
wp
Call Digium. They're ~$60 each.
Sean
-Original Message-
From: Wilson Pickett [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 10:25 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] TDM400P - upgradable how?
Hi,
I ordered the * Developers Kit which I though would be able
On Mar 10, 2004, at 9:19 AM, Stephen Davies wrote:
On Wed, 10 Mar 2004, Fran Boon wrote:
Patch failed - this is what this output is showing.
As Matt said the patch needs modifying to patch cleanly against the
current version of the code...
You didn't read his mail properly.
Steve
Thanks
Tim Sailer wrote:
On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote:
Simon,
Do the GS phones support stutter tone as-well-as
the message light?
I'm not Simon, but yes, they do. At least my -100 does. The display
backlight flashes, and you get the stutter dialtone.
Tim
My
On Wed, Mar 10, 2004 at 07:37:03AM -0600, Jim Sneeringer wrote:
I have been told that Voicetronics cards are not supported by Asterisk, but
I don't know for sure.
I used to use the VPB4 cards.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
don't think so but I wish it did
- Original Message -
From: Abdul Hakeem [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 12:53 AM
Subject: RE: [Asterisk-Users] From 0 to PBX in 2 hours
Hello,
Does anyone know if a GUI for Asterisk exists ?
Regards,
Abdul
Hello-
I'm considering some TE405P's for a customer of mine. This is the 5 volt
version of the TE410P.
Digium is now shipping these - does anyone have production experience with
these cards in the field?
Many thanks in advance.
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto,
what is this vb 206/?
- Original Message -
From: Abdul Hakeem [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 5:27 AM
Subject: RE: [Asterisk-Users] From 0 to PBX in 2 hours
Hi,
Yes, and many thanks for the offer.
Can you email or arrange for FTP ?
Cheers,
do you need a spicial phone line to run capi cards?
- Original Message -
From: Diego Ercolani [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 5:47 AM
Subject: [Asterisk-Users] Fax support and 'f' DTMF tone extension
Hello,
probably is a feature what I'm asking
On Wed, 2004-03-10 at 17:02, Chris Lee wrote:
My backlight is flashing and I have the stutter tone, only I dont have
any mail waiting, why can I not get the phones to stop doing this?
Check that
/var/spool/asterisk/voicemail/default/MAILBOXNUMBER/INBOX
does not contain a msg.txt file.
Hi Mieria,
Mireia Munoz de jesus wrote:
Hi!
When I try to call from a SIP phone to a PBX phone I get this error:
chan_oh323.c [1004] Couldn`t call 483377839
and if I get the messages from SIP debug, I have a 403 message. The
configuration of my system is:
SIP Phone ASterisk
Apologies if this is a dupe. I haven't seen my post in archives or echoed
back to me.
I'm trying to set up my first phone, Windows Messenger running on an
attached PC.
It's talking to the server, but the server never responds. Instead, it
gives this error:
Mar 10 06:26:20 WARNING[163851]:
I am no Asterisk mystro (a newbie really)
but here is my pennys worth..
I have GS budgettones..
extracts from conf files..
sip.conf
[2000]
type=friend
username=2000
host=dynamic
dtmfmode=info
[EMAIL PROTECTED]
context=sip
callerid=2000
secret=password
canreinvite=no
extensions.conf
[sip]
;local
I'm having the same symptoms using X100P's, TDM400P's and WinFax, and have
and have had no luck correcting it. I don't have a standalone fax machine to
test with.
Does anyone know if this a problem whenever faxes are sent and received with
a modem, or is it specifically WinFax? Is there any other
Simon Chappell wrote:
I am no Asterisk mystro (a newbie really)
but here is my pennys worth..
I have GS budgettones..
extracts from conf files..
sip.conf
[2000]
type=friend
username=2000
host=dynamic
dtmfmode=info
[EMAIL PROTECTED]
context=sip
callerid=2000
secret=password
canreinvite=no
Rich,
In real world, using real tool, getting real number. You don't expect to
either talk only mode or listen only mode. Per call must have Rx Tx for
inbound outbound.
The numbers look more like 83Kbits/sec (thanks Andrew Gillham) in the wiki
page or the cisco bandwidth consumsion (thanks
On Tuesday 09 March 2004 18:14, Derek Bruce wrote:
Well, after thinking about it some more... try this:
exten = s,1,ChanIsAvail(SIP/2001IAX2/[EMAIL PROTECTED])
exten = s,2,cut,ToDial=${AVAILCHAN},1
exten = s,3,Dial(${ToDial},20)
The correct syntax for Cut is:
exten =
What codecs are you using? You should be able to get quite a few speex
channels or even a load of 729 or gsm channels down your 256K.
Mark
WipeOut said:
Simon Coles wrote:
--On Tuesday, March 2, 2004 9:49 am + Steve Kennedy
[EMAIL PROTECTED] wrote:
That's the crunch (1.5/512) ...
I think this is specific to WinFax. I have my setup similar where I have
X100 - TDM400P - faxmodem(Intel 144) - Win2000 Server running its
FaxService.
I have the faxmodem plugged into a el cheapo faxmachine(for sending only,
crappy for receiving) which is plugged into the TDM400P. I have the el
It can also be due to the fact that the GS phone can't get in touch with
the time server.
Make sure you have no voicemail. Check the sip debug to make sure
message waiting is set to no.
Then its probably the time server.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
On Wed, Mar 10, 2004 at 09:03:57AM -0800, Unavailable ID wrote:
Rich,
In real world, using real tool, getting real number. You don't expect to
either talk only mode or listen only mode. Per call must have Rx Tx for
inbound outbound.
[...]
Engineering rate is per channel but to
Jim,
For the sake of argument, let's assume something about your Asterisk setup
is introducing acoustic artifacts like crackles, pops, echoes etc into the
fax transmission. The best way to overcome something like that is to make
sure your fax transmissions use ECM error correction whenever
Well I just took a look at the TAC case and things dont look good, seems the TAC are
now blaming Asterisk for the problem but I will go through there debugs and push back,
will let you know.
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: 08 March 2004 22:09
To:
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten = ${BLAH},1,NoOp is not valid, but exten
= 1234,1,NoOp(${BLAH}) is valid.
Also Asterisk NEEDS the sending fax machine to send standard fax machine
tones (CNG, I think) for it to be detected. When you
Adam,
Does Polycom license the SIP stuff from Cisco? If not, then Asterisk may be the
culprit, because all of my Polycom IP500s exhibit the same behavior. I'm running
asterisk 0.7.1, Zaptel CVS and libpri CVS both from a few days ago, but I don't recall
having this problem a few months ago
Darren,
Thanks so much for your help.
Do you know if there is other fax software that supports ECM?
Also, why should Asterisk introduce a lot of noise? (Faxes work fine without
Asterisk or with the old phone system.) I'm using a 2.4GHz Pentium 4 with
two X100P's and two TDM400P's. Faxes are
Eric Wieling wrote:
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten = ${BLAH},1,NoOp is not valid, but exten
= 1234,1,NoOp(${BLAH}) is valid.
Huh !?! Astreisk might not support it but it seems to work fine in
my setup, show dialplan expands
On Wed, 2004-03-10 at 12:25, Michiel Betel wrote:
Eric Wieling wrote:
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten = ${BLAH},1,NoOp is not valid, but exten
= 1234,1,NoOp(${BLAH}) is valid.
Huh !?! Astreisk might not support it
I've been playing around with an AGI script that pops up a callerID window
on a user's computer and I have it working with most Asterisk calls except
for calls that a queue agent receives from a queue. Does anyone know how to
run an AGI script at the time that a call is transferred from the queue
Hi,
Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway and
calls between SIP clients (phone and soft clients) are working all right. The
only problem I have, is like I have said in my mail is between sip phones and
PBX.
Best Regards,
Mireia
PS: Someone have other
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty much instant (not detectable by humans at least). So, there
Absolutely agree,
ITU standard is 64Kbits/sec.
VoIP traffic with U-law per channel is 83Kbits/sec.
VoIP traffic with U-law per call is 166Kbits/sec [measuring bandwidth per
call]
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March
All the fax modems I have seen do send CNG tones, but I have seen some very
old standalone fax machines that do not. They used to be quite common, but
in the last few years I have only seen one fax machine that does not support
CNG.
One workaround is to send the call to the fax machine on a
What versions of Zaptel, Asterisk, and libpri?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Fraizer
Sent: Wednesday, March 10, 2004 2:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice
starts after
I believe your gatekeeper or your gateway is refusing the call. This can be
a authorization problem in the gatekeeper or codec problem in the gateway.
You need to see where your call is failing. Try to do the following:
1 - Turn on the oh323 trace in the oh323.conf file adding these lines to
You need to decide if you're going to measure both sides of the call or not.
ITU standard is 64Kbits/s. That is correct. It is a standard DS0. But,
guess what. That DS0 goes both directions so, measured bandwidth per call
is 128Kbits/s using your logic.
Only consumer grade DSL/Cable
Bisker, Scott (7805) wrote:
What versions of Zaptel, Asterisk, and libpri?
I downloaded them all at the same time from CVS. I really couldn't tell you
though off the top of my head.
John
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On 2004.03.10 10:18 Jim Sneeringer wrote:
Do you know if there is other fax software that supports ECM?
Please forgive me for butting in on this thread, but I can't resist
plugging HylaFAX.
Almost all fax software, including your WinFax 7.0, that supports Fax
Class 2, Class 2.0, or Class 2.1
On Wednesday, March 10, 2004 12:19 PM, Jim Sneeringer [SMTP:[EMAIL PROTECTED]
wrote:
Darren,
snip
Also, why should Asterisk introduce a lot of noise? (Faxes work fine
without
Asterisk or with the old phone system.) I'm using a 2.4GHz Pentium 4
with two X100P's and two TDM400P's. Faxes are
Hi,
http://bugs.digium.com/bug_view_page.php?bug_id=0001193
First of all thanks
for doing this. Now we can play with any VoIP g/w in the same level field. Being
a new user always wondered why there is no radius support in
asterisk.
Sorry for the stupid
question; Why is this in bug note.
On Wed, 10 Mar 2004, John Fraizer wrote:
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty much instant (not
I have experienced this behavior on the 7960 as well.
- Chris
- Original Message -
From: Steve Creel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 3:18 PM
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts
after ring.
On Wed, 10 Mar
Sorry for my ignorance but what is the difference between using the G.729
codec and using G.729 pass thru. In my scenario below does it consider to be
using the G.729 or using it as pass through?? Do I still need licence for
the G.729?
SIP(if using g.729) ---asterisk-h323
Mark Phillips wrote:
What codecs are you using? You should be able to get quite a few speex
channels or even a load of 729 or gsm channels down your 256K.
Mark
You always have to remember that a UK ADSL line has a contention ratio
of 20:1 if you have business ADSL or 50:1 if you have the
Steve Creel wrote:
Can you test this with an extension that goes into VoiceMailMain(). My
7960 and 7960G phones both get the first couple letters of Commedian
Mail cut off (usually ...median Mail).
Just trying to quantify the delay we're talking about...
Steve
exten = 8500,1,Answer
exten =
On Wed, Mar 10, 2004 at 11:50:55AM -0800, Lee Howard said:
On 2004.03.10 10:18 Jim Sneeringer wrote:
Do you know if there is other fax software that supports ECM?
snipped
Modems that I know of that support ECM in some fashion in Class
2/2.0/2.1 are the MultiTech 5634 V92 family
Doug Harris wrote:
First of all thanks for doing this. Now we can play with any VoIP g/w
in the same level field. Being a new user always wondered why there is
no radius support in asterisk.
RADIUS is absolutely not necessary... We have countless gateways (5300s,
Qunitum, etc) running
On Wed, 2004-03-10 at 14:13, Doug Harris wrote:
Hi,
http://bugs.digium.com/bug_view_page.php?bug_id=0001193
First of all thanks for doing this. Now we can play with any VoIP g/w
in the same level field. Being a new user always wondered why there is
no radius support in asterisk.
Dear Michael,
Does your H323 driver run T38 Fax? Also, does your H323 driver have the
capability of just proxying signal, and NOT proxying signal and media, just
like the canrevite=yes in the sip scenario? Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Wed, Mar 10, 2004 at 08:41:00PM +, Michael T Farnworth wrote:
Mark Phillips wrote:
What codecs are you using? You should be able to get quite a few speex
channels or even a load of 729 or gsm channels down your 256K.
You always have to remember that a UK ADSL line has a contention
Try the attached patch. Go to your asterisk root directory and type:
patch -p0 path_to_patch/Parking.patch
.. then rebuild asterisk.
Iain
--On Wednesday, March 10, 2004 7:43 am -0500 John Congdon [EMAIL PROTECTED]
wrote:
I have applied the patch and restarted Asterisk.
But it still
Steve Kennedy wrote:
It't not quite that simple, DSL in the UK is PPPoATM, so you need to
take into account who IP is encoded at the ATM layer etc. If you're
using a reasonable bit rate codec, you can't really expect to get much
more than 1 voice channel out of an ADSL service (assume something
Steven Critchfield wrote:
Maybe you should read and understand the comments on licensing. Maybe a
going over the licensing threads here would also be needed.
For the short story, Digium dual licenses asterisk. There is a GPL
license for those of us that don't need proprietary support, and then
- Original Message -
From: hank
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 1:07 PM
Subject: where can I get Commedian mail at?
hello where can I get Commedian Mail
at?
thanks
hank
I believe that's what Steven was saying. That if this code was to ever
be included in the asterisk source then there would need to be a
disclaimer from Derek Bruce, as well as the Freeradius dev team. He
pointed out that this keeps there from being any need for a fork, and
makes things much less
Here's my patch results:
[EMAIL PROTECTED] asterisk]# patch -p0 ./Parking.patch
patching file res/res_parking.c
Hunk #1 FAILED at 25.
Hunk #2 succeeded at 228 (offset 13 lines).
Hunk #3 succeeded at 288 (offset 12 lines).
Hunk #4 succeeded at 408 (offset 13 lines).
1 out of 4 hunks FAILED --
You're right. It's symmetric so it only takes 83Kbits/sec for u-law.
IPTraf is confusing me :-)
- Original Message -
From: John Fraizer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 11:41 AM
Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729]
You need to
Anthony,
Asterisk by default allows pass through. You shouldn't need a
license. It's only when you need to do transcoding (I.E. you need to
decompress the voice, whether it be for Codec translation, to dial out a
Zap channel, which would be ULAW or ALAW, Voicemail (still technically
codec
It is included with Asterisk.
John
hank wrote:
- Original Message -
From: hank
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 1:07 PM
Subject: where can I get Commedian mail at?
hello where can I get Commedian
Mail at?
thanks
hank
Derek Samford wrote:
I believe that's what Steven was saying. That if this code was to ever
be included in the asterisk source then there would need to be a
disclaimer from Derek Bruce, as well as the Freeradius dev team. He
pointed out that this keeps there from being any need for a fork, and
[no name provided] wrote:
You're right. It's symmetric so it only takes 83Kbits/sec for u-law.
IPTraf is confusing me :-)
IPTraf is a neat tool but, information it gives should be taken with a grain
of salt. If you're looking at general statistics, it is going to show you
the combined IN+OUT
Hey,
I am looking for corporate level solution for a conference phone to use with
an Asterisk system.
Any ideas?
Current System:
Asterisk CVS-02/25/04-18:06:52
Red Hat 9.0
3 X100P cards
3 PSTN lines
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines,
On Wed, 2004-03-10 at 15:42, Jeremy McNamara wrote:
Steven Critchfield wrote:
Maybe you should read and understand the comments on licensing. Maybe a
going over the licensing threads here would also be needed.
For the short story, Digium dual licenses asterisk. There is a GPL
license for
This is more of a feature request. I've seen that there's no configuration
that allows you to remap voicemail keys(it being compiled into the source and
all) but I'd like to request the ability to do so for replacement of existing
PBX systems. It makes the integration so much easier.
Also,
Thanks Steven
Dough
Fairly appropriate sig.
Didn't you notice my email handle :)
Well I will wait for dbruce for my specific questions on how to use it.
Thanks
D.Harris (new sig.)
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Asterisk-Users mailing list
[EMAIL PROTECTED]
On Wed, Mar 10, 2004 at 05:01:38PM -0500, Jeremy McNamara wrote:
That fact is not the problem. It the fact that there is no FORK of
Asterisk that Digium secretly maintains. This is how rumors get
started.
If memory serves, you were the one who started that rumour.
I remember you claiming
On Wed, 2004-03-10 at 15:53, Doug Harris wrote:
Thanks Steven
Dough
Fairly appropriate sig.
Didn't you notice my email handle :)
Yes, I understood what it was for. A former boss used dough for his
email when his doug address was over run with spam. It took a few of us
pointing out the
This is why disclaimers are important for those who
contribute patches. If there isn't a disclaimer, Digium can not include
it in the proprietary version of asterisk. If they can not include it in
the proprietary version, they tend to not allow it in their version of
the GPL releases so they
I have noticed that the voicemail app always keeps filenames in in a
strict numerical sequence, obviously renaming files whenever a message
is deleted. I assume that message processing depends upon this
sequencing. Does the file age make any difference in determining the
numerical ordering
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
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
On Wednesday 10 March 2004 17:00, Stephen R. Besch wrote:
I have noticed that the voicemail app always keeps filenames in in
a strict numerical sequence, obviously renaming files whenever a
message is deleted. I assume that message processing depends upon
this sequencing. Does the file age
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