Gregory Junker wrote:
folder. No supplier gets a purchase if their people are not properly
trained in e-mail communication. My employer spends quite a bit as
>
You are kidding, right? "Properly trained"? By whose standards? What
international commerce committee on email standards published the
Hi folks,
Seeing as we probably wont be seeing an FXO version of the IAXy, I am
willing to resort
to having to use SIP
Any suggestions of whats available ??
Basically let it autocall a particular number and be handled as a trunk
line.
Gary
.
___
On Mon, 15 Nov 2004, Jim Dossey wrote:
> I have a client who currently has a Toshiba PBX. We are trying to
> replace it with an Asterisk system. One of the features that they have
> on their current PBX is the ability to select a POTS line by pressing a
> button on their phones. They have 10 PO
Lex Lethol wrote:
Does anyone know if this needs any special modification to work
outside the US? I have setup my country's correct tone info and
tested thru the indication.conf file.
Question would be, where does my zaptel device get the tones expected
for the busydetect procedure? How can I modi
There are a lot of functions that we have on our PBX that are not well
supported at the moment by Asterisk. Specifically the Busy Lamp
Field/button concept. We have a large MD110 pbx that does this very well.
It is essential when trying to deploy Asterisk even in a small office
environment th
As what I could read, this patch allows users to replace the moh default
system to play something else instead of mp3.
Not my problem.
Regis
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Brian West
Envoyé : mardi 16 novembre 2004 00:16
À : 'Asterisk
I will try, but the PlayBack documentation say that Playback play the whole
sound and then go to the next extension.
I begin to think that my English is really poor.
I want to play a sound during (at the same time) the DIAL, but this sound
must start from the beginning for each call. Not "jump" i
Thanks for your help.
I solved it by switching to mysql. Since i got both of them running it was
no problem.
Jens
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
List-Petersen
Sent: Monday, November 15, 2004 11:32 PM
To: [EMAIL PROTECTED]; Asteris
Roland Zagler wrote:
I use Asterisk 1.0.2 on a RedHat Enterprise Server 3.0 (Kernel 2.4.21)
and i experienced that the memory consumption of the asterisk-process
started by the init.d-script raises continously. Now, after 3 hours of
operation (on our testing-system we have 30 concurrent connections
Régis MARTIN wrote:
As what I could read, this patch allows users to replace the moh default
system to play something else instead of mp3.
Not my problem.
He posted the solution to your problem. You did not try it. What more
can we say.
The patch he sent you to will play all mp3 files (or any
Jens Hansen wrote:
Thanks for your help.
I solved it by switching to mysql. Since i got both of them running it was
no problem.
Jens
Cool. However, we may as well fix it.
Who maintains it?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Dai
Good day all.
I have a small problem
When someone calls in from the outside,dial the extension of the
internal sip phone,and the hangs up without getting any response,the sip
phone will keep on ringing?
It show that is hangs up on the Zap/vpb channel but the connection
between asterisk and sip p
I don't understand how to get call pickup to work with asterisk.
Have I to define *8 extension in the dialplan? to what?
Have I to include something, like for parked call?
Has the stable 1.0.2 version the pickup group feature?
or I need to patch it with bristuff?
Thank you
Leandro
__
hi
when finding an ABORD entry in the queue_log, how can I know what CID
that belongs to?
roy
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Till yesterday my asterisk worked fine. Today i am not getting any incoming
calls to my asterisk box.
I dont have any problem in calling outside but only with incoming even with the
same configuration.
Can any one help me to get rid off?
<>___
Aster
We have at random times the problem that a cisco phone interacting with
the voicemail can not send any digits to the voicemail.
You are checking your messages, and suddenly pressing 7 for example or
any other number does nothing.
Hanging up and calling back usually fixes the problem.
I have not
Good day all
I upgrade my asterisk and the vpb driver to the latest
I used all my previous, working, config files over.
Every thing works well but for 1 thing,playing DTMF when making a
outbound call
If I call a external number on my phone and another pbx answers and I
have to press a number it g
Hi,
I am using stable 1.0.2 version.
When I dialedin and joined a queue which in turn calls up a group( I
have only one agent logged-in, in the group ) in agent channel, it
worked well.
Keeping this call active, I made a second call which was placed in
queue, as it is intended.
When first caller
Hi,
Trying to configure a voicemail system on
FreeBSD 4.10 + asterisk 0.9.0, I found the following
problems:
1. When I try to launch VoicemailMain from
IAX Softphone (IAXComm), asterisk generates a Segmentation fault (core dumped)
and obviosly all the system went down. It doesn't happ
Hi,
I've installed a TDM31B card "successfully" but had a few problems making
calls through it - summary is below:
o Calls cannot be placed using an analog phone
o The interrupts count value in /proc/interrupts remains at zero (see
below)
CPU0
0: 7495 XT-PIC ti
On Tue, 16 Nov 2004 11:35:32 -, Victor Alvarez
<[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> Trying to configure a voicemail system on FreeBSD 4.10 + asterisk 0.9.0, I
> found the following problems:
Download the latest asterisk versions from cvs (try a make update in
the asterisk src directory
On Mon, 15 Nov 2004, Jason Williams wrote:
After the Authenticte why not do a Playtones(Dial) this will give
dialtone
The dialtone won't stop after pressing first digit then. If course you can
have an X extension that will do a StopPlaytones but that is not a good
solution since that one cannot
On Tue, 16 Nov 2004, Tobias Jönsson wrote:
> On Mon, 15 Nov 2004, Jason Williams wrote:
> > After the Authenticte why not do a Playtones(Dial) this will give
> > dialtone
>
> The dialtone won't stop after pressing first digit then. If course you can
> have an X extension that will do a StopPlay
On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:
> I don't understand how to get call pickup to work with asterisk.
> Have I to define *8 extension in the dialplan? to what?
> Have I to include something, like for parked call?
> Has the stable 1.0.2 version the pickup group feature?
> or
Hi,
Please does anyone have any more ideas on the following: I' ve set
externip in sip.conf and that simply changes the max retries error
from the internal asterisk ip address to the external address. I
thought maybe the audio wasnt going through because the client that i
was trying to call is beh
- Original Message -
From: "Walt Reed" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, November 16, 2004 1:04 PM
Subject: Re: [Asterisk-Users] Call pickup
>
> On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:
> >
Hello,
I have installed sox-12.17.6 from sources with 2.4.27 kernel but i can't mix
audio files. (For both wav and gsm formats)
For example, when i try "soxmix filename-in.wav filename-out.wav
filename.wav". Everything seems ok. There no error messages. But the output
file (filename.wav) is empty,
Hi
I would like to make point to point SIP calls between
two digium with asterisk.
I have got the asterisk working with the digium fxs
interface but can´t get the SIP session working yet.
I have configured the sip.conf and extensions.conf
files the way that I can have calls between the
extensions
On Tue, 2004-11-16 at 00:10 -0500, Gregory Junker wrote:
[snip]
> You are kidding, right? "Properly trained"? By whose standards? What
> international commerce committee on email standards published the
> training regimen of which you speak?
[snip]
Have a look at RFC1855 (http://www.dtcc.edu/cs/
I did the following:
- chan_capi-0.3.5/Makefile: uncommented CFLAGS+=-DDEFLECT_ON_CIRCUITBUSY
- recompile asterisk + chan_capi
- added /etc/asterisk/capi.conf: deflect=0800123456 ; some 0800 test number
- in etc/asterisk/extensions.conf under [tcom-in]: exten =>
98765,1,capiCD(0800123456)
- made
On Mon, 2004-11-15 at 21:03 +0100, HÃkan KÃllberg wrote:
[snip]
> >I don't know how to configure zaptel ( /etc/zaptel.conf )
I don't think you need to do anything with zaptel.conf when you are
using ztdummy. Someone on irc mentioned that ztdummy's timing is off by
about 3%. So if you are serious a
Got another question for you guys.. my x100p is on the same line as
the rest of the phones in the house my issue is for example..
a incomming call comes in and another extention picks it up before the
* box picks it up.. so the other person will be on the phone and the *
box will still pickup the
Hello
I like to setup a dial plan where sip users have the same number as
there normal phone number.
So if sombody dial the number i like that asterisk looks first is it a
number who exsist in MySQL sipfriends if not dial out with a ZAP device.
I know that extention.conf doesn't have a programmi
Thanks for your answer Tim.
Octavio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Tim
Thompson
Enviado el: lunes, 15 de noviembre de 2004 18:39
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE: [Asterisk-Users] Problem with NAT on A
I have read the the various views on top/bottom posting and it seems to me that
the proper thing to do is:
FIRSTLY, snip as much of the original e-mail as you can,
SECONDLY, reply in-line so that your answers/points are immediately below the
original questions/points,
THIRDLY, having snipped ext
Hi,
I'm testing a snom 190 behind nat, using an external stun server.
Problem is that the phone first register itself with the internal ip
address, then checks for stun and registers again with the correct ip
address.
After the registration timeout the phone register just once with the
correct i
On Tue, Nov 16, 2004 at 01:26:22PM +0100, Leandro said:
> From: "Walt Reed" <[EMAIL PROTECTED]>
> > On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:
> > > I don't understand how to get call pickup to work with asterisk.
> > > Have I to define *8 extension in the dialplan? to what?
> > > Have
On Tue, 2004-11-16 at 08:03 -0500, Jason p wrote:
> Got another question for you guys.. my x100p is on the same line as
> the rest of the phones in the house my issue is for example..
>
> a incomming call comes in and another extention picks it up before the
> * box picks it up.. so the other pers
Roland Zagler wrote:
Hello!
Can i only use one gatekeeper in OH323? Is there any documentation about
how to use gatekeeper-ids?
You can use either one or no gatekeeper at all. The gatekeeper ID is
just a special name of the gatekeeper and it can be used during the
discovery phase to find the gateke
Matt Riddell wrote:
Lex Lethol wrote:
Does anyone know if this needs any special modification to work
outside the US? I have setup my country's correct tone info and
tested thru the indication.conf file.
Question would be, where does my zaptel device get the tones expected
for the busydetect proce
On Tue, Nov 16, 2004 at 01:56:07PM +0100, Patrick wrote:
> I don't think you need to do anything with zaptel.conf when you are
> using ztdummy. Someone on irc mentioned that ztdummy's timing is off by
> about 3%. So if you are serious about your timing why don't you get a
> X100P to make sure you h
Hi
Could anyone tell me how many lines one account can take?
If I have a Switchboard extension where all calls are routed
to with one account, how many lines can that extension take
at a time?
Cisco 7960, one account, 8 lines routed to that account?
Thanks
Doug
> I've installed a TDM31B card "successfully" but had a few problems making
> calls through it - summary is below:
>
> o Calls cannot be placed using an analog phone
>
> o The interrupts count value in /proc/interrupts remains at zero (see
>below)
>
>CPU0
> 0: 7495
On Tue, 2004-11-16 at 14:39 +0100, HÃkan KÃllberg wrote:
[snip]
> Well, if it works... Compared to the crt package??
Yes it does work. I use ztdummy myself on my hardly loaded home * box
because I am too lazy to plug my X100P back in :) The 3% off will have a
negative impact when you start to get
Having a similar issue, I modified one of the zaptel tools to create a
program I call ztplay. This simply opens a channel and plays audio it
gets on stdin or a file, or fifo.
It does not even require that asterisk is running and can run along side
asterisk as long as they are not both trying to u
It's appears to be broken in that version.
Go back to sox-12-17.5(it should work).
> Hello,
>
> I have installed sox-12.17.6 from sources with 2.4.27 kernel but i can't
mix
> audio files. (For both wav and gsm formats)
> For example, when i try "soxmix filename-in.wav filename-out.wav
> filename.
> > On Tue, Nov 16, 2004 at 11:04:59AM +0100, Leandro said:
> > > I don't understand how to get call pickup to work with asterisk.
> > > Have I to define *8 extension in the dialplan? to what?
> > > Have I to include something, like for parked call?
> > > Has the stable 1.0.2 version the pickup gro
Ben Miller wrote:
Having a similar issue, I modified one of the zaptel tools to create a
program I call ztplay. This simply opens a channel and plays audio it
gets on stdin or a file, or fifo.
It does not even require that asterisk is running and can run along side
asterisk as long as they are not
Hello,
Has anyone tried "chan_misdn" in USA?
Does anyone know "chan_misdn" support Lucent 5ess or US National ISDN-1
protocols?
I live in North America and I want to connect my BRI Phones (Tone Commander
and Lucent i2021) to the BRI Card in Asterisk. I was trying to use
"bri-stuff" S/W but th
> Got another question for you guys.. my x100p is on the same line as
> the rest of the phones in the house my issue is for example..
>
> a incomming call comes in and another extention picks it up before the
> * box picks it up.. so the other person will be on the phone and the *
> box will still
I installed the new version of asterisk
But the other probelm I got was,were are using the voicetronix cards,so if
you go and put
ignorepat => 0
exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
exten => _0.,3,Dial(vpb/1-3/${EXTEN:1})
exte
On Tue, Nov 16, 2004 at 03:22:04PM +0100, Patrick wrote:
> On Tue, 2004-11-16 at 14:39 +0100, Håkan Källberg wrote:
> [snip]
> > Well, if it works... Compared to the crt package??
>
> Yes it does work. I use ztdummy myself on my hardly loaded home * box
> because I am too lazy to plug my X100P bac
>He posted the solution to your problem. You did not try it. What more
>can we say.
All my apologies.
I will try the patch, and close the thread with the result for me.
I didn't saw anything in the description of the patch that explains what you
explain. That's why I close so quickly this propo
On Tue, 2004-11-16 at 15:45 +0100, HÃkan KÃllberg wrote:
> Ohh, sorry, rtc ( real time clock ) I mean...
I don't think zaprtc is more accurate than ztdummy. Iirc it has to do
with the fact that ztdummy doesn't generate the exact same amount of
interrupts as the Digium cards do. The difference in i
Title: Newbie - NO Problems!!!
Just received my TDM400P (TDM21B) with 2 FXS and 1 FXO. Took about 2 hours to have a working test Asterisk box with 2 phones and 1 POTS line, and that includes a bare metal installation of Fedora Core 2.
Caller*ID is also working flawlessly.
My previous test
On Tue, 2004-11-16 at 09:10, Matt Riddell wrote:
> Jens Hansen wrote:
> > Thanks for your help.
> >
> > I solved it by switching to mysql. Since i got both of them running it was
> > no problem.
> >
> > Jens
>
> Cool. However, we may as well fix it.
>
> Who maintains it?
Arezqui Belaïd seems
On Tue, 2004-11-16 at 09:19 +0100, Peter Svensson wrote:
On Mon, 15 Nov 2004, Jim Dossey wrote:
> I have a client who currently has a Toshiba PBX. We are trying to
> replace it with an Asterisk system. One of the features that they have
> on their current PBX is the ability to select a POT
On 16/11/2004 16:18 Matt Riddell said the following:
In zaptel.conf where you have the loadzone=xx and defaultzone=xx, the xx
says which indications.conf entry to use.
doesn't it pull it from the structures hardcoded into zonedata.c ? iianm,
indications.conf is only used for PlayTones().
--
Rega
Hi,
Thanks for taking time to answer.
> Not enough info in the above to hint at the problem. What linux
> distro,
SuSE Linux 8.2 2.4.20-4GB
> what does your /etc/zaptel look like,
For the TDM22B card:
fxoks=1-2
fxsks=3-4
loadzone = uk
defaultzone=uk
> zapata.conf
signalling=
On Tue, 2004-11-16 at 14:35, Ken Chan wrote:
> Hello,
> Has anyone tried "chan_misdn" in USA?
>
> Does anyone know "chan_misdn" support Lucent 5ess or US National ISDN-1
> protocols?
>
> I live in North America and I want to connect my BRI Phones (Tone Commander
> and Lucent i2021) to the BRI C
o I've tried this card in all three PCI slots but no luck
o I've tried two other TDM31Bs in a similar manner with no luck
o I've tried the same with a TDM22B and get similar behaviour
Could all my PCI slots be dead or is it likely that all 3 TDM31B
cards are
dead + the TDM22B? Any clues are hi
I think at this point it would be a good idea to contact Digiun's tech
support. I've never seen a card not generating interrupts. Digium does
provide install support for their cards, which is what you need right now.
___
Asterisk-Users mailing list
[E
So, that's how my tax dollars are spent? Outrageous, and certainly
news-worthy. Good luck fighting off CNN and the like when this leaks
out.
> -Original Message-
> From: Michael Greb [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 15, 2004 10:41 PM
> To: Asterisk Users Mailing List -
Hello Everyone, I have a client that is looking at Asterisk. Does he have to get a T1 for the system or can he use his DSL line? Thanks for the help. --Tony
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On Tue, 2004-11-16 at 09:53 -0600, Jay Milk wrote:
> So, that's how my tax dollars are spent? Outrageous, and certainly
> news-worthy. Good luck fighting off CNN and the like when this leaks
> out.
It is covered under the No Child Left Behind program under continueing
reinforcement of what shoul
DSL with decent speed (256/256 or higher) will work. The connection
speed and the chosen codec determine how many simultaneous calls the
server can support. If there's a lot of other internet traffic on that
same connection, you'll also need to implement QOS.
-Original Message-
From: Ton
Hi Guys,
Heres the situation. I have asterisk setup with a T1 PRI. All extensions are
stored in a SQL database. When a call comes in, Asterisk looks up the
terminating extension in the database and the database returns another number
or extension to forward to. Asterisk then forward the or
If asterisk is running as the asterisk user, how can I control who can
run "asterisk -rx" to issue a command to it? Right now it seems as if
only root and the asterisk user can do it, but I want to allow
additional accounts to do this as well.
Thanks,
Chris
___
El mar, 16-11-2004 a las 17:28, Chris TenHarmsel escribió:
> If asterisk is running as the asterisk user, how can I control who can
> run "asterisk -rx" to issue a command to it? Right now it seems as if
> only root and the asterisk user can do it, but I want to allow
> additional accounts to do t
Can someone point me to url for the * C api (not
CAPI) reference. I am looking for the docs for the "ast_" library.
-- Mike
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I have been using LAX and getting a LOT of busy signals, i have taken
the patch off and works fine.
Seth Remington wrote:
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote:
I am still getting a Busy message when I call in to my broadvoice
number.
Is anyone else still getting that or found a fix to
I found LAX either unreachable or non-responsive for most of yesterday. I
switch to DCA and no more problems.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 16 Nov 2004, TELUX wrote:
> I have been using LAX and getting a LOT of busy signals, i have taken
>
Have doxygen installed and "make progdocs"
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael B. Murdock
> Sent: Tuesday, November 16, 2004 10:42 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk API Docs
>
>
> Just received my TDM400P (TDM21B) with 2 FXS and 1 FXO. Took about 2 hours
> to have a
working test Asterisk box with 2 phones
> and 1 POTS line, and that includes a bare metal installation of Fedora Core 2.
>
> Caller*ID is also working flawlessly.
>
> My previous test box included a US$5 X
Good luck. Best/easiest thing to do is start reading every *.c file. Thats
what I did. You can also make progdocs but I got more info by just reading
code.
Matthew
- Original Message -
From: "Michael B. Murdock" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, November 16, 2004
Just recieved our T100P for testing PRI connectivity to Asterisk. I am using
Asterisk 1.0 and libpri 1.0 and zaptel 1.0.
I compiled/installed libpri first and had no errors.
I compiled/installed zaptel second and had no errors.
I compiled asterisk third and got the following warning:
if [ -d CVS
Great!
Where did you get the card?
What installation instructions have you used?
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Tuesday, November 16, 2004 11:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sub
Apparently, no child was left behind... That's why we can find a lot of
them here.
My point was, should you have missed it, that I personally don't care
what your posting preference is. I know what mine is, but I'm willing
to work around yours and scroll more than I would think is necessary. I
Matthew:
Not sure if this is the problem, but I usually compile in this order,
different from yours:
zaptel
libpri
asterisk
asterisk-addons
regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California & London England
www.evtmedia.com
-Original Messag
The output from lspci should have shown something like:
02:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537
and its not there. Time to call/write digium support; suspect a
bad card and probably related to pci bus, but that's a guess.
> Hi,
>
> Thanks for t
I'll stop doing it when Walsh stops posting about it:
> http://www.faqs.org/rfcs/rfc1855.html
>
(from the RFC)
"...Don't wander off-topic, don't ramble and don't send mail or post
messages solely to point out other people's errors in typing
or spelling. These, more than any other behav
Joseph wrote:
Does anyone else have this problem and if so, is there a way to correct it?
Yes, we have seen this problem too, and not just with Cisco phones (also
Polycom 500). I have not yet had any time to try to debug the problem,
though.
In our case we are using RFC-2833 for DTMF transport;
I bought a few of these from PC Connection but then when I tried to order
more, they claim the product has been discontinued by the
distributor...whatever that means.
Does anyone know of a source for these that is still shipping them?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.
So, could we just agree to read around our idiosyncrasies and go back to
paying attention to the CONTENT of a message, not its FORMAT?
Discarding messages because they're in the wrong format is equal to
discriminating against another human being based on outward appearance;
be it skin-color, religi
I upgraded to CVS Asterisk 1.0 stable
last night on 2 different servers, connected to each other via IAX2.
Voice T1/PRI -> Cisco 3640
-SIP-> Main Asterisk -IAX2-> Remote Asterisk -SIP-> Phones
All using the g711ulaw codec.
We’re now experiencing gaps in
sound, the other party
Seeing a lot of this on my main asterisk servers log files
as well:
-- IAX2/NuFone/53 answered IAX2/[EMAIL PROTECTED]/38
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/38
and IAX2/NuFone/53
Nov 16 12:27:07 WARNING[1161]: chan_sip.c:683 retrans_pkt:
Maximum retries excee
Hi,
I'm trying to register Asterisk with my sip provider
but I've a problem. Here's the log file :
REGISTER sip:sip.aquanta.com SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;branch=z9hG4bK2810ee43
From: ;tag=as2e43c573
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 105 REGISTER
User-Ag
On Tue, 2004-11-16 at 11:12 -0600, Jay Milk wrote:
> Apparently, no child was left behind... That's why we can find a lot of
> them here.
>
> My point was, should you have missed it, that I personally don't care
> what your posting preference is. I know what mine is, but I'm willing
> to work a
Hello,
I have seen a lots of references to IAX in the list
but i am not quite sure what it is. What does it mean?How does it work? Why
IAX?
I just installed asterisk and so far i am using it
for IVR and routing some H.323 call through a provider. How can i benefits from
IAX?
Thanks
Kid
Any other thoughts on this?
Thanks,
Rob
> Yup, has the IDLE in there from the beginning.
>
> Does the adsi.conf have anything to do with it? (Probably no, but I'm
> grasping.)
>
> On Mon, 2004-11-15 at 21:05, [EMAIL PROTECTED] wrote:
>> Make sure your ADSI script has an 'IFEVENT IDLE THEN' bloc
Tony,
You
are going to have to be a little more specific.
What
does he want to use Asterisk for?
What
is his current PBX doing?
Does
he have a current PBX?
How
many lines at once, i.e. How many Erlangs does he
need?
Is the SIPquest server sending the 401 Unauthorized message verbatim as
you printed it here?
I.e. is the WWW-Authentcate header broken up into several lines like that?
If so, how man spaces are actually at the beginning of each new line?
Continuation lines are allowed in SIP, but I think it's only
I have a legacy PBX that is currently connected to an T1 carrying
incoming DID channels (DTMF, not PRI). I'd like to install Asterisk
with two T1 cards in between the PBX and the telco, and use it to split
off a new block of DID numbers. The remaining (old) DIDs would need to
be regenerated by As
I have a Polycom VX7000 video conferencing system (VTC) on loan that
has SIP capability. What I'm trying to do is:
ISDN PRI -(Digium Quad T1)-> Asterisk -(net)-> VX7000
If someone calls on the number for the video conferencing system,
the Asterisk server forwards the call over SIP to
Okay, I admit most of my T1 knowledge comes from the Cisco
world so please don’t castrate meJ I have
several T1’s running point-to-point between our Cooperate office and our
Branch offices, our service provider dose not provide Loop timing on any point
to point T1’s. So typically when I w
On Tuesday 16 November 2004 12:46 pm, Steven Critchfield wrote:
> On Tue, 2004-11-16 at 11:12 -0600, Jay Milk wrote:
> > I'm a fairly reasonable person, and I have yet to see one good argument
> > (and quoting netiquette is not on argument, that's opinion) for
> > bottom-posting. To me, it is terr
Hi,
Thank you for your answer.
Unfortunately, pedantic does not help.
Here's an ngrep dump of the corresponding packet.
U 212.3.244.8:5060 -> 123.123.123.123:5060
53 49 50 2f 32 2e 30 2034 30 31 20 55 6e 61 75SIP/2.0 401 Unau
74 68 6f 72 69 7a 65 640d 0a 56 69 61 3a 20 53t
I am getting the following message.
Nov 16 13:38:48 NOTICE[000640]: indications.c:397
ast_unregister_indication_country: Removed default indication country
'us'
Nov 16 13:38:48 WARNING[1087277760]: acl.c:148 ast_append_ha:
sip.broadvoice.com not a valid IP
Nov 16 13:38:48 WARNING[108727776
Thanks Dawson,
12.17.5 version works fine but i have got to mix files in gsm format first
before converting it to wav by using sox. But soxmix doesn't work directly
with wav files.
Altus, try using groups in zapata.conf. This can be done by adding
group=1
callgroup=1
pickupgroup=1
before channe
Title: Re: [Asterisk-Users] Newbie - NO Problems!!! - System Info
Rich Adamson <[EMAIL PROTECTED]> Wrote:
>
> It would be very helpful to some others on this list if
> you could
> post what type of system/motherboard you are using. The
> more detail
> the
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