11-10-2004 there was a subject:
Re: Where did USE_SIP_MYSQL_FRIENDS go?:
on asterisk.user list.
>All db specific code has been removed from the code in favor of the
>currently-in-development "RealTime" method of configuration from
>database.
>You are most likely not using the 1.0 stable branch.
>
Hello Scott,
SL> Does that include FP hardware? I don't believe that any of the
SL> PDA Xscales do, I assume that at least some codecs need FP for
SL> compression; without floating point hardware, it's going to be
SL> really slow.
1st in Xscale is integrated Micro Signal Architecture (MSA), t
A mention of it hit slashdot a little while ago.
http://developers.slashdot.org/article.pl?sid=04/11/25/2140242&tid=215&tid=126&tid=8&tid=106
Edward Beheler
Director of Information Technology
Cass County, IN
On Fri, 26 Nov 2004 08:06:34 +1300, Matt Riddell
<[EMAIL PROTECTED]> wrote:
> Matt Ridd
Lee Lee wrote:
Hi all.
i need to find someone to fix my *.
you can get more information from http://211.24.146.13/intro.txt
services will be paid.
please conact me so we can start work next week monday.
rdgs
lee lee,
are you based in malaysia? you can give me a call at 03-2094-1261 or my
cellphone
Hi all.
i need to find someone to fix my *.
you can get more information from http://211.24.146.13/intro.txt
services will be paid.
please conact me so we can start work next week monday.
rdgs
_
Download ringtones, logos and picture me
Hello,
We are working on a new Asterisk installation and
have run into some problems related to playing MusicOnHold for a caller when
they have been placed on hold by an agent, that took the call from a
queue.
A. When pressing the HOLD button on SNOM 190 and
Grandstream BudgeTone SIP pho
Leo Salas wrote:
I am just learing some Linux and have been able to setup Asterisk
samples and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1
analouge phone connected to port 4 of Digium TDM-400 with appropriate
cards installed
john drayton fule wrote:
My sound card works fine, but i can't hear the hear the person m
talking to on kphone.
based on the information i have (which is practically none :) it might
be more of a configuration file issue rather than anything to do with
your soundcard.
can you provide more detail
> > What Linux players support WAV format?
>
> Many support wav. Few support GSM codec in wav format. You can use sox
> to convert to signed linear wav so you can play with all wav capable
> players.
>
I think yours is the correct answer. The file is GSM codec in wav
format. After converting i
Hi all,
Device Information:
DEVICE: VT8233/A/8235/8237 AC97 Audio Controller
MANUFACTURER: VIA Technologies
DRIVER: snd_via82xx
regards
John Drayton C. Fule
Jr. Systems Engineer
Imperium Technologies Inc.(Philippines)
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Asterisk-Users mailing list
[
Hi everyone,
IAX2 softphone is not working with the Unregistered part in the asterisk
(chan_iax2.c and data_pgsql.c)
But with the Xlite softphone the unregistered worked properly and ast_data
properly updated the IP address and port number in the database.
I have seen some codes in the chan_i
My sound card works fine, but i can't hear the hear the person m
talking to on kphone.
On Fri, 26 Nov 2004 13:00:23 +0800, el Flynn <[EMAIL PROTECTED]> wrote:
> john drayton fule wrote:
> > Hi all,
> > Device Information:
> >
> > DEVICE: VT8233/A/8235/8237 AC97 Audio Controller
> > MANUFACTURER:
john drayton fule wrote:
Hi all,
Device Information:
DEVICE: VT8233/A/8235/8237 AC97 Audio Controller
MANUFACTURER: VIA Technologies
DRIVER: snd_via82xx
er.. what exactly is the problem again? can you please be a bit more
descriptive?
flynn
___
Asterisk
On Thu, 2004-11-25 at 21:26 -0700, Joseph wrote:
> [snip]
> > > I'm using Linux. Though I'm not sure if it is GSM file renamed to WAV
> > > or WAV
> >
> > Don't forget there are two formats can be specified. I think one is WAV and
> > one is WAV49 One works on windows, and the other doesn't.
On Fri, 2004-11-26 at 15:26, Joseph wrote:
> [snip]
> > > I'm using Linux. Though I'm not sure if it is GSM file renamed to WAV
> > > or WAV
> >
> > Don't forget there are two formats can be specified. I think one is WAV and
> > one is WAV49 One works on windows, and the other doesn't. Perhap
[snip]
> > I'm using Linux. Though I'm not sure if it is GSM file renamed to WAV
> > or WAV
>
> Don't forget there are two formats can be specified. I think one is WAV and
> one is WAV49 One works on windows, and the other doesn't. Perhaps you
> could try the 'other' one and see what happens.
On Fri, 2004-11-26 at 14:35, Joseph wrote:
> On Fri, 2004-11-26 at 12:27 +0900, Hermann Wecke wrote:
> > Joseph wrote:
> > > After somebody records a message asterisk notifies me and encloses the
> > > WAV file. Though I'm not sure if this is a WAV format. I can not play
> > > it.
> > > How to pl
On Fri, 2004-11-26 at 12:27 +0900, Hermann Wecke wrote:
> Joseph wrote:
> > After somebody records a message asterisk notifies me and encloses the
> > WAV file. Though I'm not sure if this is a WAV format. I can not play
> > it.
> > How to play received message?
>
> Did you try to use Windows Mer
Joseph wrote:
After somebody records a message asterisk notifies me and encloses the
WAV file. Though I'm not sure if this is a WAV format. I can not play
it.
How to play received message?
Did you try to use Windows Merdia Player?
In other hand, if you are receiving a .GSM file, you can use the j2
After somebody records a message asterisk notifies me and encloses the
WAV file. Though I'm not sure if this is a WAV format. I can not play
it.
According to the file specification it is:
msg.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000
Hz
How to play received message?
--
i have had some problems with the H323 channel ...
Other party not anwsering SIP 2 H323 bridge.
the chan_oh323 solves the problem. Use
it.
(Even though it is quite complicated to install but
READ the README file)
Nahuel that should solve it!!
Kido
_
I'm not saying that it would compromise *'s 'PBXness'. But you are
comparing products that have DECADES of development and maturity,
building on basic features that * is just now getting stable, and that
use proprietary hardware to accomplish these features.
Kinda my point. I reiterate, if someo
Ok, thanks!
On Thu, 25 Nov 2004 13:00:24 -0500, mattf <[EMAIL PROTECTED]> wrote:
> It is best to go through the SCRATCH_INSTALL that is listed on the project
> website:
>
> http://astguiclient.sf.net/
>
> MATT---
>
>
>
> -Original Message-
> From: john drayton fule [mailto:[EMAIL PRO
Andrew Kohlsmith wrote:
On November 23, 2004 05:28 pm, Adam Hart wrote:
iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state
--state NEW -j DNAT --to-destination ASTIP
iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE
Any reason why you need both these statem
build your server with one of these and then run a
cable to your car's lighter.
http://store.ituner.com/ituner/pw1220atxmip1.html
- Original Message -
From:
Todd
Lieberman
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, November 25,
Have a
look at http://www.twacomm.com/Catalog/Model_PF-6A.htm
As for
T1/E1, you have a big business, get a decent UPS and a
generator.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Duane
CoxSent: Thursday, November 25, 2004 10:18 AMT
I have the same problem since this morning apparently it is down
On Thu, 25 Nov 2004 11:38:15 -0700, Brian Weaver <[EMAIL PROTECTED]> wrote:
> Anyone having problems with Stanaphone registration today? I'm getting
> the following..
>
> Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_
administrator tootai a écrit :
[...]
wrapendpoint.cxx:916: no matching function for call to
`H323AudioCodec::IsDescendant (const char *)'
make[1]: *** [wrapendpoint.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.0/wrapper'
make: *** [subdirs_build] Error 1
[EMAIL PROTECTED] aste
Solution found!
I connected a recording device to the line and called the same number (near London) from different
devices. The recording I played into an audio editor (cool edit) and played the files with a program
that could decode DTMF signals. The program then showed the numbers I dialed. Exc
It is just do it.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of TELUX
> Sent: Thursday, November 25, 2004 1:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] allow=SLINR
>
> wh
HI,
1. Make sure you are running asterisk with the command
asterisk
With no arguments.
2. Make sure you are booting to run level 3 so that X-windows isn't running.
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TELUX
Sent: Thursday, Nove
On Thu, 25 Nov 2004, Colin Anderson wrote:
> I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
> opinions:
>
> 1. asterisk -p == renice -20 ??
The -p option sets asterisk to realtime priority if possible. This is
different from the traditional unix nice levels. A program
I've just got a Snom 190 phone with which I'm really pleased. I can get
the LEDs on the keys to light in response to an extension being in use
which is cool, but there's a feature I'd like to implement.
I'd like to light one of the LEDs on the keys to light if I have
night-service enabled. My Nig
I forgot to give the particulars:
Fedora Core 2
asterisk-1.0.2
spandsp-0.0.2pre6
GigaByte Triton GA7N400-Pro2 MB with AMD Athlon 2800+ CPU, 512Mb.
Cheers,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
__
I'm having an intermittent problem with fax transmissions, something I
call the streaking mascara syndrome.
TxFax will be part way through a transmission when something fails--on
the fax printer, part of the page will have correctly printed, but the
rest of the page is "streaked" to the bottom.
That was a common problem in some cases. It depended on apps,
hardware and software.
A quick FYI: I do not suggest that you use SELinux installed and operating
when running Asterisk. Fedora now comes with SELinux. Why make things
more complex.
Brandon
> Redhat 9 is running 100% cpu usage. I ha
Can't compile either
I have the oh323 v 0.6.4
I downloaded both openh323 Janus v4 and pwlib Janus v4 on inAccessNet
website
untared it, applied the patch. and compile both pwlib and open h323 (this
version compiles much much faster than the other version that seems VERY
weird to me
I believe i spe
On Nov 25, 2004, at 12:37 PM, Gregory Junker wrote:
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend,
Merlin Magix, Partner, and their new IP based PBXes support it, as do
those from Mitel, Nortel, InteCom and e
Carmi Weinzweig wrote:
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised.
Redhat 9 is running 100% cpu usage. I had a couple boxes doing this.
upgraded to Fedora and its ok.
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http:
Matt Riddell wrote:
Hi,
Just thought I'd let everyone know of our latest interview - this time
with Mark Spencer - the creator of Asterisk.
Heh, maybe I'll let you know where it is too:
http://www.sineapps.com/news.php?rssid=354
--
Cheers,
Matt Riddell
Hi,
Just thought I'd let everyone know of our latest interview - this time
with Mark Spencer - the creator of Asterisk.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily
why is allow=SLINR not an option?
im trying to use it in the iax.conf file.
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Well, it works for me .. YMMV.
Yesterday I had a problem where I had a meridian talking to * via a PRI
card, and from * to the pstn via an isdn30 link. The problem was that if the
number was bad, or engaged then the meridian line simply dropped, not giving
the operator any indication of what occur
Anyone having problems with Stanaphone registration today? I'm getting
the following..
Nov 25 11:35:58 NOTICE[229390]: chan_sip.c:4053 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
-- Got SIP response 500 "Internal Server Error" back from
216.128.82.18
___
Norman Zhang wrote:
Cannot open /dev/dsp: file or directory not found
That means you probably don't have a soundcard configured. I don't
have one in my test box either, but that doesn't prohibit asterisk
from starting up. it just means you can't do certain things from the CLI.
After stop chan_oss
I would like you to name one PBX that does not support this behavior?
Every system from Avaya including their Definity, Merlin Legend, Merlin
Magix, Partner, and their new IP based PBXes support it, as do those
from Mitel, Nortel, InteCom and every other system that I have ever
used. A typical
Thanks Kido for the answer. I have not been able to make it work yet.
My config files are:
;h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=all
dtmfmode=rfc2833
gatekeeper=200.123.148.17
AllowGKRouted=yes
context=h323
[devgw]
type=h323
e164=100
context=h323
;extension.conf
> I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
> opinions:
Bear in mind I come from a FreeBSD background. Linux might behave
differently.
> 1. asterisk -p == renice -20 ??
Why? If you have other things running on the machine, get a dedicated box
for Asterisk. It m
On Nov 20, 2004, at 11:05 PM, Gregory Junker wrote:
Most customers don't want to be in a new era. They want something
they are
accustomed to. I don't need any more impediments to making money than
I've
already got. So if the customer wants a busy lamp, I am going to do my
best to give it to them.
On Nov 21, 2004, at 11:15 AM, Wayne Sheppard wrote:
Tracy R Reed wrote:
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
I'm not surprised. Asterisk is a PBX, not
On Thu, 2004-11-25 at 11:02 -0700, Colin Anderson wrote:
> I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
> opinions:
>
> 1. asterisk -p == renice -20 ??
Unless you have done something not very smart like putting a DB on your
asterisk machine, reniceing asterisk isn't g
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
opinions:
1. asterisk -p == renice -20 ??
2. I've turned off swap with no apparent ill effects. Can anyone commment on
long term effects with moderate load (say, 30 SIP phones / 2-3K calls /day)
3. Can anyone comment on us
It is best to go through the SCRATCH_INSTALL that is listed on the project
website:
http://astguiclient.sf.net/
MATT---
-Original Message-
From: john drayton fule [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 25, 2004 5:02 AM
To: asterisk mailing list
Subject: [Asterisk-Users] astGU
Michael Devenijn wrote:
Well for example : use SIP on your LAN an use IAX to connect the outside world
...
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED] namens Bryan Mannos
Verzonden: do 25/11/2004 10:04
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
CC:
Onderwer
I am just learing some Linux and have been able to setup Asterisk samples
and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1
analouge phone connected to port 4 of Digium TDM-400 with appropriate cards
installed to dial out on.
Cannot open /dev/dsp: file or directory not found
You are right. I don't have a sound card in this box. It's suppose to
be PBX. ALSA is started though.
You do not need any sound card if you don't want to use the console
channel drivers. Just take a look at your /etc/asterisk/modules.conf and
be
Hello,
Does anyone here have DIDs for 514 area code
- Montreal, QC, in or around the 591 NXX?
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320 ext 2010
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On Thu, 25 Nov 2004, Rich Adamson wrote:
> > However, zttool reports card as "Internally Clocked". No matter how I've
> > tried, I cannot get card 1 to clock from the external source:
> > Sync Source:Internally clocked
> >
> > First span on card 0 is configured just the same:
> >
> > sp
Thank you very much for the answers - I have hooked up a special adapter and active loudspeaker on
each of the three BT lines, but when I got a line and dial a number I cannot hear any other digits
than those I dial - I would have expected something like seven DTMF bursts/digits (16662xx) before
> Problems
>
>
> - Choppy voice on calls between channels of card 1.
> - Even worse on calls between card 0 and card 1.
> - Card 0 behaves well.
> - IRQ misses for card 1. Have tried different interrupts. Same thing.
> - HDLC overrun messages on console for card 1.
Almost sounds like the
> > > > How can i get asterisk to still load, after a module has failed to load.
> > > > Can i skip over some modules.
> > >
> > > Depends on the module. Some modules are very important and can't be
> > > skipped. If it is not a module you care about, in the modules.conf, put
> > > a noload=module_
Rodney,
thanks for your question. You don't actually have to configure voicemail passwords per extension,
but per voicemail box. This is done in the config file 'voicemail.conf'.
Here you define each voicemail box number, an associated password and a name (and potentially other
parameters). When
Andrew Kohlsmith wrote:
It's not the amount of traffic that I'm concerned about, it's having a 200MHz
processor processing the encryption for a half dozen or more VPN tunnels --
passing traffic isn't an issue, it's all the processor use for encrypting and
decrypting. :-)
That's exactly what I r
Andrew Kohlsmith wrote:
seriously -- in a list with hundreds of messages a day do you really expect
that a subject line of "NEED HELP!!" will get the attention it deserves??
Sure, it will get _exactly_ the attention it deserves: none. :-)
___
Asterisk-Us
I've been running Asterisk for months with no problems. I have grown to
the point where I need an aditional TE cards. After many attempts I was
able to add the second card without affecting the performance of the
first. However, the second card is not working properly.
Setup
=
- Running on
>We use several Dell 2650 servers. Order
them with the dual DC power supply option.
> Buy a row of -48 batteries and a
-48 power source, your servers will stay up for hours.
That's only half of the solution. How will the phones be
powered? Some thoughts:
-If yo
> I am supportive of the asterisk, but I have some concern, though the
> concern also applies to traditional pbx as well. Hope someone can shine some
> light into it. Thanks.
> During a power failure situation, analog pstn lines that connect directly
> to the analog phones will most likely
i was looking but i dont find how do this:
configure the password for the extensions
read the messages
and some other things related with this
can some bady help me with some material or a explicit example.
thanks in advance
Rodney Acosta Coya.
___
As
I'm trying out ASTCC. I set the card length to 10, and generated a test
card. 10 digits. I set the extensions file to:
exten => 9175954700,1,Answer
exten => 9175954700,2,DeadAGI(astcc.agi)
exten => 9175954700,3,Hangup
I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits.
H
hi all,
this message is destinate to users who rub libunicall on asterisk succefuly
...
my first question is: what is the version of linux you use to make asterisk
run with the new libunicall? and what is the version of asterisk?
my problem is as follows:
with version FC3 I have success to ru
On Thu, 2004-11-25 at 12:21 -0300, Francisco Seratti wrote:
> > On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote:
> > > How can i get asterisk to still load, after a module has failed to load.
> > > Can i skip over some modules.
> >
> > Depends on the module. Some modules are very important
>Well for example : use SIP on your LAN an use IAX to connect the outside
world ...
Yes, I'll second that need.
-Michael
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Al Escasa wrote:
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
What I meant was that inband DTMFs do not wor
We use several Dell 2650 servers. Order them
with the dual DC power supply option.
Buy a row of -48 batteries and a -48 power source,
your servers will stay up for hours.
- Original Message -
From:
TinKoon
To: 'Asterisk Users Mailing List -
Non-Commercial Discussio
> On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote:
> > How can i get asterisk to still load, after a module has failed to load.
> > Can i skip over some modules.
>
> Depends on the module. Some modules are very important and can't be
> skipped. If it is not a module you care about, in the
If this is to gain knowledge a good source of background information is the
IP telephony cookbook http://www.informatik.uni-bremen.de/~prelle/terena/,
you should find some answers there.
One solution you did not mention is the use of a 3rd party VOIP-PSTN/PLMN
gateway - ie. you connect using H.323
On Thu, 2004-11-25 at 09:59 -0500, Giovanni Powell wrote:
> How can i get asterisk to still load, after a module has failed to load.
> Can i skip over some modules.
Depends on the module. Some modules are very important and can't be
skipped. If it is not a module you care about, in the modules.co
You just need to do something like
exten => _9.,1,Dial(Zap/g1/1666$CALLERIDNUM${EXTEN:1})
You can also do some useful translations like
exten => _9[2-8]XX,1,Dial(Zap/g1/1666$CALLERIDNUM0113${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the exte
On November 22, 2004 11:48 am, Kevin P. Fleming wrote:
> The CPU is only a limitation for a VPN if the pipe the VPN is running
> over is large/wide. These devices are typically used at the end of a
> DSL/cable connection, with a maximum bandwidth of a few megabits per
> second. I don't think that a
On November 23, 2004 08:17 am, WipeOut wrote:
> Please can someone look at my last two posts and try and shed some light
> onto why my system is dropping calls..
>
> If I don't get it right we will be forced to drop Asterisk which I
> really don't want to do..
If your last message had the same kin
On November 23, 2004 05:28 pm, Adam Hart wrote:
> iptables -t nat -I PREROUTING -p udp -d EXTIP --dport 4569 -m state
> --state NEW -j DNAT --to-destination ASTIP
>
> iptables -t nat -I POSTROUTING -p udp -d ASTIP --dport 4569 -j MASQUERADE
Any reason why you need both these statements instead of
How can i get asterisk to still load, after a module has failed to load.
Can i skip over some modules.
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ht
In my oh323.conf, i am using:
userInputMode=TONE
Is everyone trying to say that i have no hope using
oh323 when using inband DTMFs? is this problem of
asterisk? the protocol? the codec? i wish there is
still some kind of workaround.. =(
I also set inBandDTMF=yes (am not sure if that helped
but no
I GREPed the Asterisk 1.0.2 source code last night and only found a few
references to
AST_SPOOL_DIR which indicates that a patch would be rather easy.
So I'll try to do this now and share the patch. I'm thinking of something like
an option in the
Makefile called DID_HASHING, which will enable th
Pete,
I am also in the UK and I have added an include in my extensions.conf for
the file listed bellow.
exten => _15X,1,Dial,${TRUNK}/BYEXTENSION
exten => _147X,1,Dial,${TRUNK}/BYEXTENSION
exten => _NX,1,Dial,${TRUNK}/BYEXTENSION
exten => _01.,1,Dial,${TRUNK}/BYEXTENSION
exten => _07.,1,Dial,
reiserfs, ext2, ext3, etc, etc all blow up eventually, although at differnet
capacities.
Therefore simply changing from ext3 to reiserfs is, IMHO, a total band-aide
since it too has
limitations.
Hashing hundreds-of-thousands of directories seems to be to only real
alternative to keeping the
"l
Hi,
Is there anybody using Voicetronix openswitch 6 or 12 with Asterisk
???
Can interchange solutions and experiences?
Thanks
In advance
Prof. Marcelo Kruk
--
Prof. Marcelo Kruk - System Manager & Webmaster - Voip Consultant
Colegio Nacional Jose Pedro Varela - Colonia 1637 CP 11200
Phone:
Yes, as long as your service provider or your own server supports IAX2
protocol... Any comments from anyone?
IAXy currently supports IAX or IAX2? The specs say IAX, it didnt mention
about IAX2, so, is there a newer version of it or there's a way to
upgrade the firmware? Or they dont make a diff
How and what and where?
Sorry I'm a bit new to asterisk and programming
Thanks
Altus
el Flynn wrote:
Altus Snyman wrote:
Good day all
We have a voicetronix openline4 card
If someone calls in from the outside the pstn and into the system and
hangsup asterisk does not deteck the hangup
any Idea why
On Thursday 25 November 2004 12:08 am, Norman Zhang wrote:
> >> Cannot open /dev/dsp: file or directory not found
> >
> > That means you probably don't have a soundcard configured. I don't have
> > one in my test box either, but that doesn't prohibit asterisk from
> > starting up. it just means you
On Thu, 25 Nov 2004, Ashling O'Driscoll wrote:
> So basically if I want to support approx 100 calls, I would have to
> purchase a digium PRI card and then pay eircom (or whoever my service
> provider is) approx 3000 a year for the PRI ISDN connection??
100 simultaneous calls would require 4 E1 ba
Altus Snyman wrote:
Good day all
We have a voicetronix openline4 card
If someone calls in from the outside the pstn and into the system and
hangsup asterisk does not deteck the hangup
any Idea why
if i'm not mistaken the OpenLine4 cards do not have hardware hangup
detect capability -- you've got t
On Wednesday 24 November 2004 08:27 pm, Scott Laird wrote:
> On Nov 24, 2004, at 4:18 PM, [EMAIL PROTECTED] wrote:
> > Is anyone succesfully running Asterisk behind verizon residential DSL?
> > I seem to
> > be having some problems with my Asterisk server switching to Verizon.
> > I'm
> > attemptin
Thank you very much for the answer! I think it is a good path to look at. I have had a look through
our paperwork for the present pbx, and I found one document that seemed to indicate we have to dial
1666
to give the extn info to the telco. The paper is a bit old (1999) and since then we have cha
More info on this!
I bought a Grandstream budgetphone today. The buzz is not apparent
with this phone!
If I dial 8500 from the Tecom phone, it tells me its connected to voicemail
and then the BUUZZZ, which does not stop! It ignores my DTMF tones for
entering the mailbox, just BUUUZZZ.
If I do the
Hi,
When I call from a SIP phone to X-lite client on a PC (on same local network)
I see the following error:
-- Executing Dial("SIP/home2-a189", "SIP/xlite1|20|tr") in new stack
Nov 25 12:45:01 WARNING[1589264]: chan_sip.c:600 __sip_xmit: sip_xmit
of 0x973a0bc (len 765) to 192.168.1.27 returned -
There were some old libs from another installation in
/usr/lib/asterisk/modules. "rm"ed them and "make install"ed them again,
worked.
Alex Mack wrote:
Hi everybody!
I've managed to compile the bristuff patch on asterisk from
Junghanns.net. I want to run this on the quadBRI card built into the P
Hi Jasko:
Kphone and Linphone can do v6. We've experienced problems with voice
quality using these in the wide area. Not sure this is something to do
with playout buffering or other packet handling problem (?).
We have also developed a prototype UA based on RAT and the Vovida
sipset. It was demon
hi all,
this is the first time i write to this mailing...I hope to do not
wrong and that to speak an understandable English.
in these days i have the problem to record calls of a particular
extension according to database entry that i can change in every
minute.
when a call arriving to Asterisk, i
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