Re: G.729 codec on Yellow Dog Linux for various PPC
Kristian Kielhofner [EMAIL PROTECTED] wrote:
This is probably a good time to ask if there is any
planned support for a g729 binary for YDL and
G3/G4, etc. I
I try to get the following to work:
Sipgate.de and sipgate.co.uk are configured as gateway, while the
ATA-186 has two phone sets attached.
I tried:
ATA settings as described at:
http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt
(just with a fixed IP)
sip.conf:
==
[general]
[EMAIL PROTECTED] wrote:
Folks,
I am trying to determine the best way to allow a station to monitor the
status of another station.
For example:
a reception set needing to see the status of 20 or 30 phones
OR
an executive assistant wanting to have appearances of several other
extensions, in order
Ronald Wiplinger schrieb:
[sipgate.de]
type=friend
username=5552220
secret=my_password3
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.net
should be fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=my_password4
BetaTeilchen wrote:
Ronald Wiplinger schrieb:
Thanks for helping me, but it still does not work.
[sipgate.de]
type=friend
username=5552220
secret=my_password3
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.net
should be fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
Maybe you should start reading here:
http://www.voip-info.org/wiki-Asterisk+introduction to get basic
knowledges of Asterisk
Ronald Wiplinger schrieb:
BetaTeilchen wrote:
Ronald Wiplinger schrieb:
Thanks for helping me, but it still does not work.
[sipgate.de]
type=friend
username=5552220
Hi list,
I'd like to have comments from the bristuff / QuadBRI users, others are
welcome to as I'm really lost and need to move on.
I have the following setup: a first asterisk is connected to the legacy
Alcatel PaBX to connect to a remote site with a second asterisk server.
Todd Lieberman wrote:
Wo trevor, Format and start over? Don't go crazy, just remove the files
created by make install.
Fighting for weeks to get a more-or-less stable telephone system can
drive a man to do extraordinary things like rebuilding a server from
scratch!
We are making process,
On Sat, Oct 23, 2004 at 06:40:12PM -0600, Michael Loftis wrote:
--On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff
[EMAIL PROTECTED] wrote:
Look for support by whatever operating system you plan on running.
I second thatpretty much any P4 based hardware should be
I would use a Western Digital Raptor SATA Harddisk, also gives you a performance boost of your system + it aint that expensive as scsi.
And my dream setup for asterisk would be:
dual xeon, intel xeon motherboard, 2gig ram for each cpu and a few raptor or scsi disks + some wildcard digium
On Sat, Oct 23, 2004 at 09:29:00PM -0500, Carlos Chavez wrote:
I am installing a new * server using Fedora Core 2 but I ran into a
problem after I installed the X100P. When FC2 boots it runs KUDZU to detect
new hardware and it detected the card and insists on loading the module
crc_ccitt
A customer has ordered some voice prompts from Digium's TheVoice
online store. They say the recordings' sound was good when they
listened to it on their Windoze boxes. However, then Asterisk is
playing back the recordings, the volume is far too high and they sound
really bad. This is particularly
Things that are productive.
1. I am sure there are free programs that will allow you to adjust the
files to sound more like the originial recordings as well as converting them
to gsm. Do some searching and learning and fix it yourself. Mail Digium
directly so that they are aware of the problem
Option #4, send me the files and $100 via paypal and I will fix them for
you.
- Original Message -
From: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 6:37 AM
Subject:
Hi,
I have just set up an Asterisk box.it sure is a big job to get
everything perfect, especially when you have picky users.
Anyway, the box has 2 X100P's and a couple of sipura spa-2000's
connected to the LAN.
1 of the lines connected to the X100P's goes straight to extension 1000
after a
A flex grow is like a channel bank. A normal PRI comes into a router. The
router breaks out some channels for data and the other voice channels become
analog POTS lines. You will need POTS cards.
I am positive that you could have your T100P and asterisk provide this
function so that you
hi
would it be hard to implement CallerPres support in chan_sip?
roy
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May be risky if your email is screwy but it solves your problem
Add: delete=yes in your voicemail.conf.
- Original Message -
From: Stephen R. Besch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 11:55 AM
Hi,
I've some problems compiling/installing the ZAPHFC-Driver. I've download the actuell
version bristuff-0.1.0-RC4a from junghanns.net. I use SUSE 9.1 with kernel 2.6.5.-111.
I've made the symbolic link to Linux-2.6 and test the link successfully. I've done
make oldconig, make menuconfig
Hi Steve,
On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro
[EMAIL PROTECTED] wrote:
1. I am sure there are free programs that will allow you to adjust the
files to sound more like the originial recordings as well as converting them
to gsm.
that's all very cool, but if you read my post
Hi, I have just been able to compile asterisk, so that says that I'm fairnly new to Asterisk. I'm still figuring out how to use it with my User Agents. My requirement is:1. To make a "Group" containing some agents (SIP User Agents) as members.2. To start a conference between the members of the
You are supposed to be able to either press flash or quickly push the actual
hook switch.
- Original Message -
From: Nikhil Jogia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 7:54 AM
Subject: [Asterisk-Users] Call Waiting
Hi,
I have just set up an Asterisk
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
A customer has ordered some voice prompts from Digium's TheVoice
online store. They say the recordings' sound was good when they
listened to it on their Windoze boxes. However, then Asterisk is
playing back the recordings, the volume
However, the format the customer ordered was WAV, whereas all the
included recordings are of course GSM. Has anybody had similar
experiences? I tried to convert the WAV files to GSM using sox but
since I don't know what parameters are best in this case, the results
weren't satisfactory. Any
Hi Benjamin,
I looked at NuFone.net and some others, but it appears that
IAX is not right for my system.
I'd say this is only because you don't know enough about IAX yet ;-)
[Many comments explaining how IAX would work wonderfully if all my
VoIP hardware were replaced with IAX-compatible
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
On Sun, 24 Oct 2004 07:47:17 -0400, Steve Totaro [EMAIL PROTECTED] wrote:
1. I am sure there are free programs that will allow you to adjust the
files to sound more like the originial recordings as well as converting
them
to
Hi folks,
I have upgraded asterisk from 0.8 to 1.0 on my gentoo server and it
won't start now. It crashes on random points while loading the modules
somewhere between res_crypto and chan_iax2
the last messages are either:
===
Yuck! Error
Helpful URLS about SOX/wav/gsm
Have you seen these?
Converting:
http://www.voip-info.org/wiki-
Convert+WAV+audio+files+for+use+in+Asterisk
Volume:
http://www.voip-info.org/wiki-Asterisk+sound+files
Other bits and bobs:
http://www.marko.net/asterisk/archives/0212/0384.html
e.
On 24 Oct 2004, at
Did you upgrade zaptel and libpri before upgrading asterisk?
- Original Message -
From: Tomas Carnecky [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 9:44 AM
Subject: [Asterisk-Users] random crash at startup
Hi folks,
I have upgraded asterisk from 0.8 to
When I try to compile asterisk-oh323, errors as following:
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory `/root/willis/asterisk-oh323-0.6.3b/wrapper'
./check_ver /root/willis/pwlib pwlib
./check_ver /root/willis/openh323 openh323
gcc -shared
Steve Totaro wrote:
Did you upgrade zaptel and libpri before upgrading asterisk?
do I need zaptel?
I have libpri-1.0.0 but no zaptel installed.
in the gentoo ebuild the dependecy is like thik:
DEPEND=virtual/libc
media-sound/mpg123
dev-libs/newt
doc? ( app-doc/doxygen )
hi
I'm currently using sipfriends from asterisk-stable and I've enabled
MYSQL_USERS as well. Are mysql/odbc/whatever _users_ available in
extconfig yet?
roy
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On October 24, 2004 07:47 am, Steve Totaro wrote:
2 . If you dont want to go through all of that, kindly ask Digium to have
the files fixed for you. I seriously doubt they have their own sound stage
and most likely outsource this type of business. Chances are the people
they outsource the
On October 23, 2004 10:58 pm, Michael Loftis wrote:
mmm... any packaging is better than none. I regularly destroy things on
systems when it's not been put into proper packaging because we upgrade the
system, and there's no record of something being installed, nor what it
depends on, so it
would it be hard to implement CallerPres support in chan_sip?
There is support for outgoing calls, but this patch breakes incoming
callerid:
http://bugs.digium.com/bug_view_page.php?bug_id=0002471
Greez
Andreas
_
Listen to music
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
A sound stage is a facility used to create and process professional
recordings. They can be used by anyone employed by an company.
- Original Message -
From:
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
A sound stage is a facility used to create and process professional
recordings. They can be used by anyone employed by an company.
http://www.theivrvoice.com/
would
On October 24, 2004 10:24 am, Steve Totaro wrote:
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
Oh I dunno, for telephone IVR you don't need much of a sound stage. Convert a
bathroom into one with a lot of insulation,
On Sun, 24 Oct 2004 14:47:56 +0100, Elliot Moore
[EMAIL PROTECTED] wrote:
Helpful URLS about SOX/wav/gsm
Have you seen these?
[snip URLs]
yes, I have played with those and all I did achieve was making the
recordings worse, but thanks anyway.
However, it seems now that this is not a common
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
http://www.theivrvoice.com/
would seem to imply otherwise. I'd be a bit surprised if any company had
enough work to keep her employed full-time, so the works at Digium line
sounds a bit fishy to me.
I think when
get the new versions of libpri zaptel and asterisk and install them in that
order. should work.
- Original Message -
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 10:05 AM
Subject: Re:
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
Folks,
I am trying to determine the best way to allow a station to monitor
the status of another station.
For example:
a reception set needing to see the status of 20 or 30 phones
OR
an executive assistant wanting to have appearances
Hi Benj,
On Sun, 24 Oct 2004 23:39:06 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
[snip URLs]
yes, I have played with those and all I did achieve was making the
recordings worse, but thanks anyway.
However, it seems now that this is not a common problem so I have to
On Sun, 24 Oct 2004 15:27:53 +0200, Stewart Nelson [EMAIL PROTECTED] wrote:
No, I don't want to replace existing gear.
fair enough.
It would be expensive
that I don't agree with, especially not if you do it yourself, but anyway.
There are other reasons, too. For example, the Cisco 827-4V
Steve Totaro wrote:
get the new versions of libpri zaptel and asterisk and install them in that
order. should work.
I have libpri-1.0.0 and now I've reinstalled asterisk-1.0.0 but it still
doesn't work, and there is no Digium hardware in the server so I don't
need zaptel.
tom
I have installed the first time Asterisk, (forgive me simple questions)
I have also installed the demo.
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
On Sun, 2004-10-24 at 05:10, Tzafrir Cohen wrote:
One obvious solution is not to automatically load kudzu.
chkconfig --remove kudzu
Another obvious solution of the same sort is modprobing the zaptel
module earlier in the boot process.
I can't seem to figure out , though, where kudzu
On Sun, 24 Oct 2004 12:01:19 -0300, Nicolás Gudiño [EMAIL PROTECTED] wrote:
I've never ordered from thevoice, but I have converted some MP3
I guess you mean WAV
to gsm
and after fighting with sox parameteres I came up with this:
sox in.wav -t gsm -r 8000 -g -b -c 1 out.gsm resample -ql vol
Yep, same here with Ericsson T610
Reason: AT+BRSF is not implemented in Ericsson Cellphones.
Kind regards,
Martin List-Petersen
http://www.marlow.dk/
On Wed, 2004-10-20 at 22:20, Jon Radon wrote:
Running Asterisk CVS-HEAD-10/19/04-04:34:45, just tested with my Sony
Ericsson T68i. Couldn't
just try it.
- Original Message -
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 11:05 AM
Subject: Re: [Asterisk-Users] random crash at startup
Steve Totaro wrote:
get the new
Ronald Wiplinger wrote:
I have installed the first time Asterisk, (forgive me simple
questions)
I have also installed the demo.
I solved it with the newest cvs version !!!
bye
Ronald
After testing demo (call 1000, call 600, ...) I changed in the
extensions.conf:
; include = demo
include
Just an idea, couldnt you remove the zaptel hardware, run kudzu and remove
the hardware module via kudzu then disable kudzu again?
- Original Message -
From: Carlos Chavez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday,
Steve Totaro wrote:
just try it.
I installed the old 0.9 version of asterisk and now it works, even with
libpri-1.0.0.
I've found out thet the kernel module ztdummy wasn't loaded while I
tried to start asterisk, could this have been the problem?
tom
more info on ztdummy and zaptel i am sure will solve your issue.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
- Original Message -
From: Tomas Carnecky [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday,
Just tried the patch you made with the latest CVS and it patches
fine although it does not work. Now when I hit # it does not
send the DTMF to the other side at all. Although hitting ##
does get the transfer. Now # doesn't do ANYTHING :)
I'm not sure why that is, it works with all our
99% of the companies I call that say hit # after entering your
response, doesnt actually require the #.. I've only encountered a few
places that if you don't hit the # it ignores your response,
eventually sending you to an operator or hangs up on you.
On Sun, 24 Oct 2004 09:37:09 -0700, Randy
If this is call waiting on the CO line, I found to flash the CO line you
have to (flash *0) to answer it. If it is another station calling your
phone while you are on , a normal flash will do.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nikhil Jogia
Hummm.. thats cool..
I havent tougth about being re-provisioning the iaxy box :)...
But how do you detect the dns change? wich ddns company are u using?
Jim Van Meggelen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wilson Pickett
Sent: October
ah good thinking, i didnt even factor CO call waiting into the equation
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 1:19 PM
Subject: RE: [Asterisk-Users] Call
clearly * is receiving the #, as ## does do a transfer. so why
is a single # not being sent onward as dtmf?
I noticed on X-Lite that # in a dialstring is sent URL-encoded or similar,
and Asterisk doesn't understand it.
Could this be something similar? Perhaps sip debug will reveal?
Steve
On Sat, 2004-23-10 at 19:43 -0500, Me wrote:
Any chance you can pass me the Beta Version or let me know how to get it
myself?
I'm sorry, I can't distribute the beta. You can ask Uniden support -
although I doubt they'd hand it out willingly. I wouldn't expect much
more of a wait before an
Benjamin on Asterisk Mailing Lists wrote:
On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
http://www.theivrvoice.com/
would seem to imply otherwise. I'd be a bit surprised if any company had
enough work to keep her employed full-time, so the works at Digium line
Looks like what you want is not music on-hold, but rather a streaming
server
On Oct 22, 2004, at 4:23 PM, Ryan Courtnage wrote:
On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote:
On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
[snip]
Is there a way to force MusicOnHold() to be
Why not just create a context that plays static msgs whenever someone
is transfered thereThank you for calling Monthly special etc
...
then transfer them back when the person at the biz picks up
On Sun, 24 Oct 2004 14:23:04 -0400, Emilio Panighetti [EMAIL PROTECTED] wrote:
Looks like
Geotel is a company that Cisco bought which provides call control across
geographically dispersed locations. The simplest application is being
able to query call queue status at another location. For example, a
call comes in and can be sent to one of three call center locations.
Geotel can
in order to get the cid from the spa3k to *, i need to turn on
PSTN / PSTN-To-VoIP Gateway Setup / PSTN CID For VoIP CID: = YES
this produces a sip invite as follows:
Frame 1 (1092 bytes on wire, 1092 bytes captured)
Ethernet II, Src: 00:90:69:6d:e8:00, Dst: 00:30:48:80:b3:72
On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys
[EMAIL PROTECTED] wrote:
after a couple of days work banging my head against the wall (bloody standards
my arse), i've got chan_bluetooth to a point where it's starting to function
- certianly more than just proof of concept now.
Will
Hi,
Anyone could use Asterisk Prepaid with a MySQL database? Thanks.
Nahuel Ramos.
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Hi,
I've been running * for a couple of month now. However, now i want to
run ztdummy. Compiling works (apart from some warning regarding
strict-aliasing), however installation gives missing Unresolved symbols:
if [ -d /etc/modutils ]; then \
/sbin/update-modules ; \
fi
depmod: ***
Hi all,
I am running the Asterisk with CVS-HEAD-10/25/04.
When I type reload in console, whatever the incoming/outgoing sound
volumn becomes very loud until I stop the asterisk and restart it.
It's running no problem before I've upgrade the asterisk. Is there any
configuration I need to
I don't know but it's IMHO, this should be just the opposite. Single # for
a transfer and double ## to send the key on as DTMF. How many objects in a
dialplan start with a #?
Lyle
- Original Message -
From: Randy Bush [EMAIL PROTECTED]
To: Barton Hodges [EMAIL PROTECTED]
Cc: splatters
What if you call an external system and get a voicemail. Press # to finish
your message . you would have to press ##.
IMHO I think most users are not sophisticated enough to transfer calls. If
they are they can press ##.
Or am I missing something? :)
S.
-Original Message-
From:
I do some script type programing and have seen this in other uses. IMHO, it
would be easier to program this way. Single # go to transfer function. Get
# as first character in transfer, send out the DTMF tones instead and drop
the request to transfer.
I could be all wet on this, but my feeble
Personally I think like you... but I have to force myself to consider the
dim wits that use my PBX. :) They are fat old men who barely understand
what a telephone is... let alone VOIP. :)
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese
Sent:
Hello, working on trying to get the Iaxy setup from behind a NAT.
I have done everything the way I think it should be done but I can't seem to
get dial tone and each time the device trys to register with * I get this
message on the console:
***
Oct 24 15:15:11 NOTICE[131080]:
Get a hint! :-)
Check out the hint priority in extensions.conf. There are also some
details in the wiki.
I've looked all over the wiki, and all the documentation I could get my
hands on, Where did you find anything about the hint priority? I am
interested in trying to make this work.
Ok, all of this makes sense but I guess the bigger question is..
How does one check their voice mail and delete it by using a phone and
dialing into *? Is there a magic extension or series of buttons to push to
get someone into their mailbox?
Thanks,
Todd
--
Start Your Own ISP!
19 Question + this one and no answer;
Does anyone have a clue what causes Unknown RTP codec 72 received notice
and how to fix it?
Regards
Danny
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On Sun, 2004-10-24 at 20:08, Benjamin on Asterisk Mailing Lists wrote:
On Wed, 20 Oct 2004 13:37:28 +0100, Theo Zourzouvillys
[EMAIL PROTECTED] wrote:
after a couple of days work banging my head against the wall (bloody standards
my arse), i've got chan_bluetooth to a point where it's
On Sun, 24 Oct 2004 21:42:49 +0100, Martin List-Petersen
[EMAIL PROTECTED] wrote:
Regarding the headset, i have not seen how that works yet, but i would
say you would need to enter the number somewhere (maybe special prefix
on any phone + phoneno. to get the call to the headset)
Oh well, I
On Sat, 2004-10-23 at 01:06 -0400, Jim Van Meggelen wrote:
Few will disagree that the careful application of netiquette will be a
benefit to any newsgroup/mailing list/board; and top posting is
something that should be used sparingly. Nevertheless, top posting is
not the horrid crime some
better get to reading.
Basically you need to create an extension that points to voicemailmain.
- Original Message -
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 4:30 PM
Subject: Re:
-Original Message-
From: Danny Froberg [mailto:[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 4:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unknown RTP codec 72 received
19 Question + this one and no answer;
Does anyone have a clue what causes Unknown RTP codec
http://www.voip-info.org/tiki-searchresults.php?words=voicemailwhere=pages
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 5:27 PM
Subject: Re: [Asterisk-Users] Re:
From what I read about a year ago was that it was a carrier hosted
solution that actually controlled the ss7 switching at the exchange
(basically no call costs from tromboning, and was only implemented into
an ip-centrex or hosted call centre application.
Are you saying that enterprises can buy
I bought out IAXy devices from NetXUSA, who also sent me this short
installation document which I've copied below:
Quick Start Guide for Digium IAXY Device
Determine the IP address of each unit by viewing the logs on
your DHCP Server to see which IP address your IAXY has taken, you
Thanks Steve, it's not that I have not been reading (ask my wife how many
nights I have slept in the last week), and it's not that there is not a huge
amount of info out there. The problem I am having is finding the info I need
in any sort of organized way.
The searches I do sometimes come up
[EMAIL PROTECTED] wrote:
Get a hint! :-)
Check out the hint priority in extensions.conf. There are also
some details in the wiki.
I've looked all over the wiki, and all the documentation I
could get my hands on, Where did you find anything about the
hint priority? I am interested in
What Manfred wants to do is not that uncommon. I've used the method
that William has suggested in the past. On a lot of corporate phone
systems this is a simple option in the programming. Another way is to
simply advertise your specials over your music on hold and repeat
them... Hoping that the
I found in Google a h323.conf file, but not on my Asterisk installation.
Do I need to do more than h323.conf ???
I have a h323 phone and would like to replace it as one connection to my
Asterisk,
Thanks for your hints.
bye
Ronald
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
Has anybody tested any gateways from ACT:
http://www.act-tel.com.tw/Index2.htm
They have four different configurations:
4xFXS - 4xFXO
2xFXS - 2xFXO
1xFXS - 1xFXO
4xFXS
I emailed them but they didn't bother the respond.
--
#Joseph
___
Asterisk-Users
vi /usr/src/asterisk/channels/h323/h323.conf.sample
vi /usr/src/asterisk/channels/h323/README
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, October 24, 2004 8:01 PM
Subject:
Okay, I have removed the IDE Controller and am now using onboard. The
problems below still exist--
Trevor Peirce wrote:
show translation still reveals the iLBC column in the 8700 to 9600
range though. LPC10's row is also in the 900s.
show translation recalc 10 still causes the * console to
On Sun, 2004-10-24 at 10:24, Steve Totaro wrote:
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
A sound stage is a facility used to create and process professional
recordings. They can be used by anyone employed by
On Sun, 2004-10-24 at 17:52, dean collins wrote:
From what I read about a year ago was that it was a carrier hosted
solution that actually controlled the ss7 switching at the exchange
(basically no call costs from tromboning, and was only implemented into
an ip-centrex or hosted call centre
I am buying a Snom phone this week. I will play with this feature and see
what I can get going. I will share my findings.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Van
Meggelen
Sent: Sunday, October 24, 2004 6:05 PM
To: 'Asterisk Users Mailing
Hi all,
I am trying to slap together a script that will email2sms the details of
the voicemails left on my * box to my gsm phone. I can't figure out how
to get my script to pick up the voicemail vars like ${VM_MSGNUM},
${VM_DATE}, ${VM_MAILBOX}, ${VM_CALLERID}, ${VM_DUR}. Right now I have
this:
hello list.
looking for a way to have several FXS ports on an asterisk box, lets say
oh... 300, just for shoots and giggles. would i need special telco
equipment? if so, what kind? i already have a 23 inch cabinet, which i'm
told telco equipment uses 23 inch. any insight on this would be greatly
On Sun, 2004-10-24 at 22:02 -0400, James Dumais wrote:
hello list.
looking for a way to have several FXS ports on an asterisk box, lets say
oh... 300, just for shoots and giggles. would i need special telco
equipment? if so, what kind? i already have a 23 inch cabinet, which i'm
told telco
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
You seem to not realize that those who are knowlegable are only so due
to the vast amount of time we put into learning. I'm sure there are many
people who are like me and are trying to spend a lot of time learning
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