. Fleming wrote:
[EMAIL PROTECTED] wrote:
Just swiming around in it here.. Any thoughts? It seems to me that
you MUST use something like MGCP or H.248 to connect the call to the
PSTN (media gateway) since the specific DS0 to be utilized will be
included in the ISUP messages..
No, you can just do what
Robert Augustyn wrote:
Hi,
I want to reset IP600 to the factory settings but when I press 468* and hold
it asks for password?
Is there another way?
Robert
Hmm,
On the IP500, I believe this is where you enter the MAC address. You
know, I'm pretty sure this question is all over the mailing
el Flynn wrote:
What should I be looking for when I'm testing the unit? Anyone can
offer some hints as to what I need to look out for?
I know voice quality should be something to look at, especially when
all channels are in use. What else should I be looking out for?
Blah blah.. specs.. Post a
Thanks for your reply Lubo .
If any user could provide any feedback on simultaneous call
volumes they have seen on * with sip -h323 transcoding
it will be highly appreciated.
i could not find any information in the archives.
Regards,
John
-
HI John,
It also depends which
Hi! All
I would appreciate if someone could advice me on how stable is sip-h323
h323-sip translation as well as how many calls can it handle when doing such
translation.( assuming single 2.8Ghz intel processor 1GB RAM)
Regards,
John
--
Christopher wrote:
Thanks guys, really appreciate the responses.
Actually I've tried the suggestions in this document with absolutely
no luck at all unfortunately, and turning off fixup protocol udp sip
was the key to allowing my remote phone to ring to an internal phone
(when fixup is on I
Henry Devito wrote:
www.thirdlane.com http://www.thirdlane.com has already written a
close dsource webmin module. I have no idea how much it costs or how
well it works.
I've attempted to contact thirdlane to get pricing on their GUI and
can't seem to get anyone to reply.
My personal
HiA followup on my previous problem (no sound) description: I compiled zaptel and ztdummy, and loaded themThen i recompiled asterisk and configured sip clients and a conference. When i load the ztdummy module into the kernel, and run asterisk, the conference room seems to work, but i cannot hear
Joe Greco wrote:
911 Testing is a very complicated issue. For a clec it typically
involves scheduling with them so they will expect your call. Also we
frequently use false addresses (that are MSAG resolvable) and some very
sophisticated PSAPs even have fake addresses that MSAG resolve to a
911 Testing is a very complicated issue. For a clec it typically
involves scheduling with them so they will expect your call. Also we
frequently use false addresses (that are MSAG resolvable) and some very
sophisticated PSAPs even have fake addresses that MSAG resolve to a
testing ESN.
Hi
I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1
The process of installation was the following: First I compiled and installed
Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the
ztdummy (modprobe ztdummy) and then i installed Asterisk:
make
make install
Hi all,
I am sure there is a way to get a Multitech MVP110 working with * in
H.323 mode. I have just not been able to figure out how from the MVP110
side. Could someone please share their config setup with me for the
MVP110 and the * side??
TIA,
Robert Webb
lights up correctly.
Any ideas what could be the problem?
--
Richard Cook
[EMAIL PROTECTED]
705-497-9320
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Hi All,
I'm trying to record a phone call.
I'm using the Monitor command with the m flag for a SIP to SIP call.
I'm running:
Asterisk CVS-HEAD-12/17/04-16:55:26
One side of the call is significantly quieter than the other. Am I doing
something wrong??
Thanks,
Brett
lookup, you're looking
for lidb access..
-m
On Sun, 26 Dec 2004, Lyle Giese wrote:
That's good to get a general idea, but number portability only tells you
which carrier has the block. It does not let you know about specific
numbers :-{
Lyle
- Original Message -
From: Matt Klein [EMAIL
Sigh..
This shouldn't be so hard. Ok guys, I'm trying to figure out how to
support end user features for my users. Perhaps some of them are typical
verticle service features like *69, *72, *66, etc, you get the picture.
Here's my deal. Sure implementing them one by one is easy enough. But
Karl Brose schrieb:
The configuration file is 'phone.conf'
mode=dialtone
[interfaces]
mode=dialtone
format=slinear
echocancel=medium
device = /dev/phone0
This is my phone.conf but I don't get a dialtone
I have the ixj driver running and a cat /proc/ixj after asterisk
start tells me one reader and
Karl Brose schrieb:
The configuration file is 'phone.conf'
mode=dialtone
[interfaces]
mode=dialtone
format=slinear
echocancel=medium
device = /dev/phone0
This is my phone.conf but I don't get a dialtone
I have the ixj driver running and a cat /proc/ixj after asterisk
start tells me one reader
So for newby users of SJPhone... can you tell us exactly what goes in
what box to connect to a standard AsteriskPBX using the latest
interface. I've had no luck so far.
thanks...
On Wed, 2004-12-08 at 09:40, Girish Gopinath wrote:
Hi,
--- Norman Zhang [EMAIL PROTECTED] wrote:
I'm following
;-)
James
Subject:
Re: [Asterisk-Users] Re: Ethernet Channel Bank idea
From:
Michael Graves [EMAIL PROTECTED]
Date:
Wed, 15 Dec 2004 06:52:33 -0600
...SNIP...
I think that one real opportunity, perhaps of many potentials, is the
smaller installation. We suffer the lack of small format, reliable
first wrote that message ;-)
Thanks;
James
Subject:
Re: [Asterisk-Users] Voice Prompt Info
From:
Christopher Dobbs [EMAIL PROTECTED]
Date:
Fri, 10 Dec 2004 16:24:00 -0800
You should not put the press or the number in the prompt
generic.
[EMAIL PROTECTED] wrote:
I am looking for titles that fit into the string:
press 1 for the DEPT department or press 1 for DEPT
but if you have other suggestions, let me know.
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, You have reached the accounting department, etc...
Thanks;
James
Subject:
Re: [Asterisk-Users] Voice Prompt Info
From:
Christopher Dobbs [EMAIL PROTECTED]
Date:
Sat, 11 Dec 2004 13:07:07 -0800
Your previous messages came
list
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solution for me? :)
-Brett
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Steve Frank wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, November 30, 2004 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] National (US) callerid name
On Mon, 29 Nov 2004 23:52:29 +, Corvin [EMAIL PROTECTED] wrote :
Thank you very much now it's much easier to read.
Kind regards,
Corvin
I suppose in light of some people's discontent with the idea of diverse
sources of information they never say what would end these conversations
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You could do a pri+101 on the dial.. so if the first dial fails, try the
second
Original Message:
-
From: Luke Connolly [EMAIL PROTECTED]
Date: Fri, 19 Nov 2004 15:42:27 +1100
To: [EMAIL PROTECTED], [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Best SIP phone for high quality telemarketing
I'm really happy with my Polycom IP 600
http://www.polycom.com
://www.thesuperweb.gr Website ìå ÁóöáëÝò Controlpanel áðü 6 Euro êáé äþñï
ôï domain óáò!
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Actually we have used RedHat Enterprise AS in the military has well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Tuesday, November 02, 2004 3:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
the BRI installed, so I can't post configs etc.
TIA.
Jon
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I agree I would like to know what has changed from 1.0 to 1.0.2
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Thompson
Sent: Tuesday, October 26, 2004 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED
WARNING[-154464]: loader.c:429 load_modules: Loading
module app_realtime.so failed!
Ouch ... error while writing audio data: : Broken pipe
Ouch ... error while writing audio data: : Broken pipe
It crashes now and won't start.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
??
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hi,
how do u prevent unauthorized usage or block users temporarily to use Asterisk
services ?
Is defaultip and secret enought ? what u do to prevent this.
tia
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I have tried that on the GrandStream Budgetone phones and the transfer
does not work on them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Monday, October 25, 2004 2:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
hmm... now tring.. somone to know how can I redirect the output of the sip debug
into file 'cause it is really hard to grasp (several pages is just one call)
On Fri, 22 Oct 2004 01:24:24 +0300, raptor [EMAIL PROTECTED] wrote:
On Fri, 22 Oct 2004 01:36:29 +0900
Benjamin on Asterisk Mailing
hi,
Can someone point me to a list of a common numbers used for different
functions ex. callparking,forwarding etc...
I can thin of my own but want to know is there some standard wich is
good to follow.
tia
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i've browsed the the voip site... but cant find is there a way the asterisk server
to cut the ECHO, or it is only possible with hardware solution ?
I'm connecting 2 voip phones.
tia
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of the phones then I have the situation as before one
can hear the other
but not the other way around.
tia
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Do you have a list of those providers that use IAX?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
on Asterisk Mailing Lists
Sent: Friday, October 22, 2004 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
tia
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, only the software. Does anyone have the part
number for software upgrades? (the $8 one referenced on the Wiki)
Thanks,
~chris
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are not detected properly by Asterisk and not transmitted to the
other party. Any ideas?
Chris
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/ .
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Having used both at ISP's I can say though ATT is a better quality
service, and more responsive to trouble, their sales cretins are absolute
liars and shouldn't be trusted any further than you can throw them. if
they represent something, GET IT IN WRITING,SO YOU CAN SUE THEM, as they
are simply
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 7:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Mandrake 10, Request for comments.
My * is presently running fine on Mandrake 9.2, but Ive been
entertaining
moving to Mandrake
discriminate against older applicants)
At 11:11 7/7/2004, you wrote:
Let them suffer like the record industry. Times are changing, better change
with them or fall by the wayside.
- Original Message -
From: Joe Baptista [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 06, 2004 5:19
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into it.
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and context, which are useful in
an Asterisk environment. That's pretty much all that comes to mind at the
moment.
Mark
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,
Brian
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off list and let's see if we can isolate the issue. Can't
tell from the words you've used what steps you've gone through to date.
Rich
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and never get to actual content about
it. often google yields better results.
randy
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: iJKLmNoP
Context: VPWS
Example extensions.conf:
[general]
static=yes
writeprotect=no
[default]
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
[voicepulse-incoming]
; This context tells Asterisk what to do with
; incoming calls from VoicePulse (if you have signed
; up for DIDs
;
; We
Interesting point. Are there VoIP terminators that can accept iLBC or where
can I find them? Are there any hardware endpoints that can handle this
codec? Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Viernes, 14 de Mayo de
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ronald R.
McDaniel
Sent: Viernes, 14 de Mayo de 2004 02:56 p.m.
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] OMG THE SKY IS FALLING!! NOT!!!
I sent the following to the great Mr. Louderback:
Mr. Louderback,
If I was your friend, I would
/500,0/15000
info = 950/330,0/1000
dialout = 500
Sergio Serrano Revuelto
Avanzada 7
Original Message:
-
From: Jose Maria Guisasola [EMAIL PROTECTED]
Date: Tue, 6 Apr 2004 22:52:16 +0200
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] indications.conf settings for spain
Somebody has
me should more
information be needed.
Kind regards,
Devon H. O'Dell
mail2web - Check your email from the web at
http://mail2web.com/ .
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It seems that older version of asterisk does the codec negotiation fine.
I have one machine running CVS-12/19/03 and this can negotiate codec
g729 and gsm fine.
The newer version cvs-1/27/04 does not negotiate codec correctly. The
ougoing connection can only go either g729 or gsm.
--
David
but.. not getting to connect SIP-IAX2 and the problem is not
only with VoicePulse but with another provider as well in the same
situation, GS(SIP)- * - IAX2 - ITSP
-- Call accepted by 66.234.228.132 (format G729A)
-- Format for call is G729A
-- IAX2[voicepulse]/2 is busy
-- Hungup
and ulaw, others. So you have to set your GS
to do ulaw and set your codec accordingly in your sip.conf as well.
--
David Kwok
FWD#/IAXTEL# : 17001813482 ext 1002
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[EMAIL PROTECTED] wrote:
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: Wed, 18 Feb 2004 09:48:08 -0600
Subject: Re: [Asterisk-Users] codec negotiation
Reply-To: [EMAIL PROTECTED]
Why do you need 729? I just called your IAXTel number using GSM and
connected
[EMAIL PROTECTED] wrote:
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED] [EMAIL PROTECTED]
Date: Wed, 18 Feb 2004 09:48:08 -0600
Subject: Re: [Asterisk-Users] codec negotiation
Reply-To: [EMAIL PROTECTED]
Why do you need 729? I just called your IAXTel number using GSM and
connected
I have set up call queue for incoming calls. However, when I try to
transfer call after answering the queue to another station, the call is
hung up. The agent login into Asterisk by AgentCallbackLogin(). When the
agent's phone rings the agent pick up the call queue.
Is it normal behaviour that
...)?
mark
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Philipp von Klitzing wrote:
You'll need to provide the CODEC that you are using in X-Lite!
The codec used in Xlite is 711uLaw. I guess it is one of the preferred
ones other than gsm. And it is of small size.
--
David Kwok
FWD#/IAXTEL# : 17001813482 ext 1002
smime.p7s
Description: S/MIME
Subject: [Asterisk-Users] meetme not working
Reply-To: [EMAIL PROTECTED]
I am trying to set up meetme functionality but am unsuccessful so far.
When I dial the extension, an announcement says, That is not a valid
conference number, please try again.
In order for meetme to work you need either
Referring to my previous post about degradation of voice quality when
having more than 2 connection.
The actual route is:
pc xlite - local asterisk box - iaxtel - local asterisk
I have tried out a different situation:
pc xlite - local asterisk box - iaxtel
and the second connection
pc xlite
Any one has documented how-tos for making voicetronix openline 4 to work
with Asterisk.
I have been contacting Australian Digium resellers and Digium cards are
not approved in Australia. So I suppose Australian users are interested
into putting Voicetronix in use.
Any expereience to share
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
The error message from the concole:
-- Executing MeetMe(SIP/1002-e9ca, 4700) in new
On Thu, 2004-01-15 at 19:18, [EMAIL PROTECTED] wrote:
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
This works
My ADSL speed is Uplink 128kbit and Downstream 512kbit.
The mii-tool does not tell whether eth0 is in full-duplexed mode. It
just say that it is 100baseTx.
David Kwok
smime.p7s
Description: S/MIME Cryptographic Signature
Hi
my question is:
which is the best distribution to work with asterisk?
thanks
mark
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?
thanks
mark balester
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Hi All
I have problem trying to receive incoming calls from iaxtel.com. The
error message is rejected connect from ip address - iaxtel.com.
I have set up the iax.conf file as follow:
port=5036
allow=gsm
register=dkwok:[EMAIL PROTECTED]
[dkwok]
type=friend
context=from_iaxtel
My
/2003
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Thanks, any special configuration requirement?
Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza
Sent: Lunes, 22 de Diciembre de 2003 05:01 p.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicetronics
iTS [EMAIL PROTECTED
Hi, where can I find info on configuring pass-through mode.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of SW
Sent: Jueves, 18 de Diciembre de 2003 11:29 p.m.
To: asterisk users
Cc: Clif Jones
Subject: [Asterisk-Users] G729 question
Hi Clif,
My
promised. This will be excellent.
No more questions .. case closed... Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tjapko Smits
Sent: Miércoles, 17 de Diciembre de 2003 11:34 p.m.
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] gateway VoIP h323
nope h323 only sorry...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Areski
Sent: Lunes, 15 de Diciembre de 2003 04:50 p.m.
To: Asterisk-Users Mailing-list
Subject: RE: [Asterisk-Users] Howto to test asterisk applications -
VoIPTesting Solution
CALLGEN
OK. I tried with 1.12.2 and indeed problem fixed. Thanks, Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Sbado, 13 de Diciembre de 2003 04:54 a.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RH9 and h323.conf
SW wrote
of
satisfied customers will kill me if this is true) Also that any written
information On h323 will become available in about 1 month or so . Maybe
this info is usefull. Thanks again for your answer, Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of SW
Thanks, I will try this tomorrow. I already have these libraries installed
on my box. KR, Tjapko.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Sbado, 13 de Diciembre de 2003 04:54 a.m.
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
mail2web - Check your email from the web at
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