Hi Guys.
I have some questions regarding how to interconect * server with each other.
We are 3 asterisk servers and we added each other as friend on iax.conf.
So far everything was working or appeared to be working fine until:
1. One of the server changes it IAX2 port each time it reboots, why
There is a program for linux called centericq, this program is for
connecting a linux box to aim, icq, msn, etc something like trillian.
Anyway, this centericq lets you define external commands that can run when
you send it a message containing certain words. Also, you can define a
system call in
with a register and a peer and user entries:
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=user
context=fwd-incoming
auth=rsa
inkeys=freeworlddialup
[fwd-gw]
type=peer
host=iax2.fwdnet.net
user=user
secret=pw
qualify=2000
disallow=all
allow=ulaw
callerid=Anton Krall
But for connecting 2
, let me know.
Anton Krall
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scheda
Sent: Domingo, 13 de Marzo de 2005 02:31 p.m.
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Text Messaging or AIM
On Sun, 13 Mar
Check out centericq on freshmeat. Lets your linux box be on MSN, ICQ, AIM,
etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jess Coburn
Sent: Domingo, 13 de Marzo de 2005 03:18 p.m.
To: Scheda; Asterisk Users Mailing List - Non-Commercial Discussion
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Text Messaging or AIM
On Sun, 13 Mar 2005 16:13:04 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
I already have this workling for remote linux admin. For example, each
linux box has it MSN user and I have them on ly MSN list. So
Firefly?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roman
Zhovtulya
Sent: Domingo, 13 de Marzo de 2005 04:49 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Looking for a free SIP/IAX softphone with IM
**' with no technology!
On Mar 4, 2005, at 7:00 PM, Anton Krall wrote:
Guys, this error has been driving me nuts and I see no indication
anywhere as to what it may mean.
Anybody has any clues on this?
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en
callerid=Anton Krall
But for connecting 2 * servers you need just this:
register = user:[EMAIL PROTECTED]
[remoteasterisk]
type=friend
language=sp
host=dynamic
secret=pw
context=mty-incoming
disallow=all
allow=ilbc
auth=md5
trunk=no
qualify=3000
accountcode=mike
Why not use
=iax2.fwdnet.net
user=user
secret=pw
qualify=2000
disallow=all
allow=ulaw
callerid=Anton Krall
But for connecting 2 * servers you need just this:
register = user:[EMAIL PROTECTED]
[remoteasterisk]
type=friend
language=sp
host=dynamic
secret=pw
context=mty-incoming
Guys.. Why is it that when a call comes to a call queue and in term gets
assigned to an agent, if that agent tries to xfer the call using # or any
other feature, it doesn't do anything? I just hear the pleeps on the phone
but asterisk doesn't intervene with the Transfer prompt.
Am I missing
Queues and Transfers
On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote:
Guys.. Why is it that when a call comes to a call queue and in term
gets assigned to an agent, if that agent tries to xfer the call using
# or any other feature, it doesn't do anything? I just hear the
pleeps
Jolio, no, I checked the wiki and didnt see that parameter
there, but I just checked show application queue and made the necessary
modifications.
Thx Guys!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of João
AmaroSent: Martes, 15 de Marzo de 2005 04:12 a.m.To:
Asterisk
On dial command yes, wtWT just in case, and it works when Im the one that
originated the call, but for example, I have the same problem that you have
when the call comes in thru a Zap channel. I cant make transfer to work
eventhough the dial command that sent the incoming Zap call to me has wtWT.
Guys.. I just noticed that my grandstream handytone 286 ata are having
problems with voice cutoffs... We can listen to the person on the zap
channel (x100p cards) without problems but they sometimes listen to us with
cutoffs.. like He ...lo. ow...r.. you and it comes and goes.. this
doesnt
Guys.. I just noticed that my grandstream handytone 286 ata are having
problems with voice cutoffs... We can listen to the person on the zap
channel (x100p cards) without problems but they sometimes listen to us with
cutoffs.. like He ...lo. ow...r.. you and it comes and goes.. this
doesnt
Speaking of this. Why is it that sometimes the port is shown as something
differente than 4569 on some hosts? For ex.
Host UsernamePerceived Refresh
State
210.80.176.12:221108990608214 1.2.3.4:4569 60
Registered
And why that host changes port
,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Thursday, March 17, 2005 4:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAX Registration being lost
Speaking of this. Why
PROTECTED] On Behalf Of Henry Devito
Sent: Martes, 15 de Marzo de 2005 10:55 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voice getting cutoff
check for interrupt conflicts, cat /proc/interrupts
- Original Message -
From: Anton Krall [EMAIL
cutoff
Anton Krall wrote:
What do you think?
CPU0
0: 16148159 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 0 XT-PIC usb-uhci
8: 1 XT-PIC rtc
10: 161351663 XT
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, March 19, 2005 5:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Voice getting cutoff
How can I change the IRQ of the cards?
-Original Message-
From: [EMAIL
Guys.
Anybody usign mysql addons for cdr? I just noticed that records are still
been sent to the master.csv file but seems not all of them.. Which makes me
think about which one has the actual truth? And why are not all records been
sent to the csv files and mysql?
Also, can I then disable the
Guys.
Which PC softphone (IAX2 or SIP) can support getting a url on the dial cmd
and opening a web page on the users computer?
Any choices?
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Guys. I know this might be a long shot but wanted to check with the gurus.
I have outlook 2003 on my computer and wanted to check if there is a way of
connecting outlook with asterisk so that caller id name could be set based
on my outlook address or contacts? Each time a call comes in for me,
Thx!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
Schmidt
Sent: Domingo, 20 de Marzo de 2005 02:17 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk and outlook
Anton Krall wrote:
I
Guys.
Im having a big problem transfering incoming calls thru zap channels to some
other extension. If the call is made by me to the outside via zap channels,
no problem, hitting # gets me the transfer prompt, but if the call comes in
thru zap and eventhough I am sending the call from the zap
I just installed tapi and some app called identapop pro. I havent tested
incoming calls yet but so far, I cant get calls out using outlooks.
I configured TAPI for asterisk inside outlooks and I set TAPI to these
configs:
TAPI connects using the manager to asterisk without problems.
As channels
OK, the outbound problem is fixed... Now, my other question is, anybody
using identapop for popup CID on your screen?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 20 de Marzo de 2005 03:34 p.m.
To: 'Asterisk Users Mailing
How do you dial 1800 numbers using FWD? any prefixes needed?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of cmisip
Sent: Domingo, 20 de Marzo de 2005 03:39 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] FWD to
15:29:18 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
Guys.
Im having a big problem transfering incoming calls thru zap channels
to some other extension. If the call is made by me to the outside via
zap channels, no problem, hitting # gets me the transfer prompt, but
if the call comes
Discussion
Subject: Re: [Asterisk-Users] TAPI
you are realy having fun this sunday :)
On Sun, 20 Mar 2005 15:42:07 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
OK, the outbound problem is fixed... Now, my other question is,
anybody using identapop for popup CID on your screen?
-Original
.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TAPI
A free solution would be to use YAC in conjunction with netcat. A guide is
on the wiki.
On Sun, 20 Mar 2005 15:42:07 -0600, Anton Krall
[EMAIL PROTECTED] wrote:
OK, the outbound problem is fixed... Now
]: app_queue.c:374 changethread: Can't
change device '**Unknown**' with no technology!
Mar 11 12:21:32 WARNING[22142]: app_queue.c:374 changethread: Can't
change device '**Unknown**' with no technology!
On Mar 4, 2005, at 7:00 PM, Anton Krall wrote:
Guys, this error has been driving me nuts and I
up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF
Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To: Asterisk Users
DISA?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh Alberts
Sent: Lunes, 21 de Marzo de 2005 01:55 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Script to Authenticate User and Dial Out
Hello. I'm looking for a script that I
Hey Guys.
Im trying to make a script but I need some ideas on the logic behind it.
Here is what Im trying to do:
I have 2 zap channels. I want to make a macro that would do some random
choices on which of the 2 zaps to use, then test if it is available, if it
is, make the call, if not, then go
Will it get added to cvs-head?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Bockman
Sent: Martes, 22 de Marzo de 2005 07:39 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Chanspy is back !
vote
Guys.
Anybody doing ChanisAvail on IAX2 channels?
Im trying to do this:
exten = s,7,ChanIsAvail(IAX2/anton:[EMAIL PROTECTED])
But I get that the chan is unavailable eventhough I can make calls to that
channel. Is there any chatch?
The channels is defined as peer and Ialso tried doing a
09:59 am, Anton Krall wrote:
But I get that the chan is unavailable eventhough I can make calls to
that channel. Is there any chatch?
The channels is defined as peer and Ialso tried doing a register on
iax.conf for that channel. Everything is registering ok and I CAN make
the call.
Just
] On Behalf Of Anton Krall
Sent: Miércoles, 23 de Marzo de 2005 09:46 a.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Chanisavail and IAX2
Yep, I use qualify also with 1000
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Sent: Miércoles, 23 de Marzo de 2005 11:16 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Chanisavail and IAX2
it doesn't work with current CVS, it works with 1.0.7
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users
I tried using that. Works for outbound calls thru outlooks but didn't find a
way to make it do the cidlookup on incoming calls, also, doesn't have any
help that worked for this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Jueves,
I tried using that. Works for outbound calls thru outlooks but didn't find a
way to make it do the cidlookup on incoming calls, also, doesn't have any
help that worked for this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry Devito
Sent: Jueves,
Any word on when CHanisAvail for IAX2 will be on CVS?
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-Asteriskdatabase(LookupCIDName)
There is a separate component that you can purchase that will allow popups
from outlook db.
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 24, 2005
Guys. Im writing this because Ive checked the wiki, Xten website and read a
lot of docs and still cant figure out a way around the NAT issues. Maybe
somebody else can give me some ideas from a fresh perpective.
My test setup is this:
Asterisk- 2wire homeportal
Guys.
Can you dial 800 and 888 toll free numbers using FWD? how do you dial them
cause I tried using 1800x and 1888x and I simply get a nobody can
asnwer the call signal on asterisk.
Can you dial 800 toll free from FWD?
___
Asterisk-Users
Guys. Im writing this because Ive checked the wiki, Xten website and read a
lot of docs and still cant figure out a way around the NAT issues. Maybe
somebody else can give me some ideas from a fresh perpective.
My test setup is this:
Asterisk- 2wire homeportal
See what I get when trying that number thru FWD
-- Executing Dial(SIP/casa1-7552, IAX2/fwd-gw/*18005551212|60|rwt)
in new stack
-- Called fwd-gw/*18005551212
-- Hungup 'IAX2/fwd-gw-2'
== No one is available to answer at this time (1:0/0/0)
-- Executing Playback(SIP/casa1-7552,
Ill do a search on that...
Besides that, how would you make * work for sip if for example this:
Softphone - nat - internet - nat - *
??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Sábado, 26 de Marzo de 2005 02:17 a.m.
To:
.
To: Anton Krall
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 800 numbers and FWD
Got posted unfinished:
Dial(SIP/casa1-7552, IAX2/fwd-gw/*18005551212|60|rwt)
Take a *close* look at that line and compare it to the one on the FWD page:
Dial(IAX2/${FWDNUMBER
Im checking the wiki for call files info and seems somebody has a wake up
script that runs call files at certain times.
Do you know if its possible to run a call file by using some other methods
different from cron jobs or at? The wiki mentions that it might be possible
to do this is you modify
-Commercial Discussion
Subject: Re: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 15:55 -0600, Anton Krall wrote:
Im checking the wiki for call files info and seems somebody has a wake
up script that runs call files at certain times.
Do you know if its possible to run a call
.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 15:55 -0600, Anton Krall wrote:
Im checking the wiki for call files info and seems somebody has a wake
up script that runs call files at certain times
at certain times
On Mon, 2005-03-28 at 16:37 -0600, Anton Krall wrote:
BTW, can you use the same call file to make 2 calls in order or just 1
call per call file?
1 call per file
What I want to do is first make a call to a sip phone and playback
some file and then make another call to the same sip
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Lunes, 28 de Marzo de 2005 10:29 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call files run at certain times
Anton Krall wrote:
Any examples that can do that?
extensions.conf
Discussion
Subject: RE: [Asterisk-Users] call files run at certain times
On Mon, 2005-03-28 at 23:00 -0600, Anton Krall wrote:
But this doesn't work in an environment where multiple person are using
it..
For example, multiple call files with multiple announcements..
IT can easily enough. Go
Guys.
Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of
ways to get around nat but I would like to hear some success stories about
handling nat users with multiple voip phones behind nat.
I have my asterisk box behind but ports are forwarded (5060 5004 1-2
for
I like your idea, Ill play with it for a while and see what comes out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Martes, 29 de Marzo de 2005 12:36 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
:[EMAIL PROTECTED] On Behalf Of Paul Fielding
Sent: Martes, 29 de Marzo de 2005 12:52 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
would
Matt.
I gave your ideas a try and made it work with a twist. Use a macro but...
Here is the good part, call the macro from a call file using application,
passed parameters like name of the sound file, telephone to call, etc.
Voila! Works great!
Thx for the hints Matt.
-Original
both in and out of NATs without
reconfiging.
No special ports being forwarded for the clients. They seem to work behind
whatever NATs we throw at them without difficulties...
later,
Paul
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non
ManxPower
Sent: Martes, 29 de Marzo de 2005 10:28 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Anton Krall wrote:
Any problems with RTP or voice just on one side?
So as long as you use some STUN
Guys.
I just finished installing a new asterisk box and here comes the first
problem.
The box doesnt have zaptel cards or anything, its a plain RH9 with asterisk.
Every compiled perfectly and when trying to run asterisk -vg
I get all the messages shown below, no errors except for a
- Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No prompt after installing
Anton Krall wrote:
Guys.
I just finished installing a new asterisk box and here comes the first
problem.
The box doesnt have zaptel cards or anything, its a plain RH9 with
asterisk.
Every compiled
-Commercial Discussion
Subject: Re: [Asterisk-Users] No prompt after installing
Anton Krall wrote:
Guys.
I just finished installing a new asterisk box and here comes the first
problem.
The box doesnt have zaptel cards or anything, its a plain RH9 with
asterisk.
Every compiled perfectly and when
List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No prompt after installing
On Wed, 30 Mar 2005 21:43:49 -0600, Kristian Kielhofner [EMAIL PROTECTED]
wrote:
Anton Krall wrote:
Guys.
I just finished installing a new asterisk box and here comes the
first problem.
The box doesnt
Ill try that one, thx!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Miércoles, 30 de Marzo de 2005 11:19 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] No prompt after installing
-Commercial Discussion
Subject: Re: [Asterisk-Users] No prompt after installing
Anton Krall wrote:
ntpdate ntp1.cs.wisc.edu
30 Mar 23:15:20 ntpdate[3840]: no server suitable for synchronization
found
:(
ntpdate pool.ntp.org
Try that.
--
Kristian Kielhofner
Guys.
While working with agents and queues, you have settings like timeout for
agents that dont answer within certain times but I have a question, if you
use autologoff, for example, setting the timeout for 15 seconds and
autologoff for 30 seconds, then, the agent wont be logged off since you
Guys.
I know how to make 2 asterisk servers dial each other via IAX and such but I
was wondering if there is a way to only have 1 centrl voicemail and not have
each asterisk have its own voicemails.
Is this possible?
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Sent: Sábado, 09 de Abril de 2005 11:27 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Servers and One Central Voicemail
Anton Krall wrote:
Guys.
I know how to make 2 asterisk servers dial each other via IAX and such
but I was wondering
Thats could be a problem
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent: Sábado, 09 de Abril de 2005 11:51 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Multiple Servers and One Central
be cumulative? For example,
set it to 20 second so that if 2 rings of 10 seconds are not answered, then
autologoff the user?
Hope you can help.
Thx!
Anton Krall
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But voicemailboxes have to exists on all asterisk servers
right?
Also, what happens if for example, the user is accessing
his VMB on server 1 and changes his password, then travel to where server 2 is
and tries to access his VMB? the config on server2 would still have the old
one so
be cumulative? For example,
set it to 20 second so that if 2 rings of 10 seconds are not answered, then
autologoff the user?
Hope you can help.
Thx!
Anton Krall
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What about MWI config on sip.conf or iax.conf? [EMAIL PROTECTED] All you users
have the same contexts for checking MWI? By context, does it mean the
dialing context or just the [] context used on the voicemail.conf file?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Guys.
I just had a weird problem. I have my Dial cmd configured with mwtWT as
parameters however, a call came in thru a zap channel and I answered on a
sip phone. I tried using # as configured on my features.conf file to
transfer the call but the transfer prompt never came in, so I asked the
: [Asterisk-Users] weird call transfer problem
Anton Krall wrote:
Guys.
I just had a weird problem. I have my Dial cmd configured with mwtWT
as parameters however, a call came in thru a zap channel and I
answered on a sip phone. I tried using # as configured on my
features.conf file
Anybody doing it with Grandstream handytone ATA 286?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Allen Niven
Sent: Miércoles, 13 de Abril de 2005 08:29 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
are probably your only bet in that case...
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Conference
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Wednesday, April 13, 2005 8:19 AM
To: 'Asterisk Users Mailing List - Non
Im reading some tips on dimentioning asterisk servers and I was wondering
about the show translation command.
The numbers shown, what do they mean? Ms needed for transcoing but... How to
measure if they are good or bad?
For example:
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729
Guys.
Anyway had problems with G BT 100 or 101 volume? Seems the volume is too
loud and when talking it makes the voice cut off due to saturation.
Anyway to reduce the input voice volume on the phones?
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Guys.
Anyway had problems with G BT 100 or 101 volume? Seems the volume is too
loud and when talking it makes the voice cut off due to saturation.
Anyway to reduce the input voice volume on the phones?
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this has been asked a lot of
times :(
__
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, another protocol and can it go behind
firewalls or do you need to configure or open certain ports?
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Anton Krall
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http
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Startup Question
Anton Krall wrote:
Guys, Im new to asterisk and voip but Im have a couple of questions
regarding the initial setup.
1. Im going to install an asterisk server at home, where I have 2 phone
lines, what kind
!
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How do you make SIP work behind NAT without having to change anything on the
firewall for example, those cable modems
So far, Ive tested this using softphones and only iaxphone has been able to
work using IAX, eye lite or something for FWD that uses SIP says it cant
connect to the provider...
Guys.. which ATA is better for connecting analog phones (features,
stability, experiences, etc)?
Sipura 2000 or Handy Tone 286, etc?
What are you experiences?
__
Anton Krall
/software/asterisk-1.0.0/channels'
make: *** [subdirs] Error 1
Any ideas what might be wrong?
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Anton Krall
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development libs are the cause.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: 18 February 2005 09:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problems compiling on mandrake
Guys.. Im having
]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: 18 February 2005 09:53 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Problems compiling on mandrake
Guys.. Im having problems compiling asterisk under Mandrake 10.1 with kernel
2.6.8.1-12mdk
-onlyperson' (language 'en')
But I dont hear anything... any experiences with this kind of errors and/or
[EMAIL PROTECTED]
Thx!
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 2:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] No Sounds
I just installed [EMAIL PROTECTED] to see how it works
without the license (I might be a little mistaken here...)
I hope this helps. I have not use [EMAIL PROTECTED], it might be different.
Let me know,
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday
Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users
-Users] No Sounds
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
This is a very good
be the normal setup on
sip.conf for this to work?
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?
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