Remove the space before Dial
On Mon, 2003-10-13 at 05:27, John Foster wrote:
Hi List..
I m getting this mesg while trying to dial an extension, both SIP UAs
are registered with asterisk, m trying to dial extension 1015 from UA
[EMAIL PROTECTED] to extension 1016 of UA [EMAIL PROTECTED]
Try http://www.fnords.org/~eric/asterisk/ It contains simplified config
files as well as other information.
On Mon, 2003-10-13 at 06:34, Conrad Braun wrote:
Hi,
could somebody name the minimum configuration files asterisk needs to
run with a SIP phone?
what do i need apart from
Where does the 4 wire change to a 2 wire?
On Tue, 2003-10-14 at 03:30, Mark Spencer wrote:
Echo has nothing to do with TCP vs. UDP. It's an analog phenomenon that
occurs where the hybrid is, where the four-wire circuit changes to a
two-wire circuit.
--
Sample configs and more:
Yes, of course. However, that would be a feature of the SIP phone, not
Asterisk, since Asterisk isn't providing the dialtone on your SIP phone,
the phone is doing that.
On Tue, 2003-10-14 at 16:28, Chris Hariga wrote:
Hi,
si posible on SIP phones to have the dial tone after 9 like on the FXS
For T-1 use a Digium card for about US$500
On Tue, 2003-10-14 at 17:02, Roger Schreiter wrote:
Hi,
my asterisk experiences with isdn cards supported by i4l
are not very good, but with avm a1 and capi everything
works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20,
german ISDN).
Now
set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf
On Wed, 2003-10-15 at 09:50, Eduardo Goncalves wrote:
Hi list,
I have a cisco 827 with 4 fxs and an * gateway, like this:
[c827]--sip-[asterisk]-em---PSTN
The codec used is g711alaw over a 9Mb
false hangups.
On Wed, 2003-10-15 at 13:29, Eduardo Goncalves wrote:
On Wed, 15 Oct 2003 11:16:03 -0500
Eric Wieling [EMAIL PROTECTED] wrote:
set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf
Thanks for the tip. Could you explain me why these options set to yes
may
There's a bug report on bugs.digium.com. Most people don't need to
transfer calls that go to outside numbers, only calls that come IN to
asterisk and calls between extensions and so most people don't run into
the problem.
On Mon, 2003-10-20 at 09:42, Louis-David Mitterrand wrote:
On Mon, Oct
Actually it requires CHANGING the SIP protocol. Asterisk already
changes the SIP protocol when you use nat=yes and many clients also
change the SIP protocol to work with NAT.
On Mon, 2003-10-20 at 11:31, Chris Albertson wrote:
Asterisk works perfectly fine in back of a NAT firewall, as long
as
show application MusicOnHold
Also show applications for more information on all the Asterisk
applications.
On Mon, 2003-10-20 at 11:13, Kevin wrote:
Is there anyway for a sip station to play MoH out of the speaker?
I know I can do it by calling the station and putting it on hold.
For
You should start by reading the specific SIP and RTP RFCs. SIP is less
of an issue than RTP (as someone else pointed out)
On Mon, 2003-10-20 at 12:47, Chris Albertson wrote:
--- Eric Wieling [EMAIL PROTECTED] wrote:
Actually it requires CHANGING the SIP protocol. Asterisk already
changes
Voicemail2 already does the date and time.
On Mon, 2003-10-20 at 22:03, Kevin wrote:
Has there been any discussion as to having asterisk voice mail play an
optional message envelope with caller ID, date and time of message?
___
Notice the mailbox= entry in
http://www.fnords.org/~eric/asterisk/zapata.conf.html
Works the same in http://www.fnords.org/~eric/asterisk/sip.conf.html
On Mon, 2003-10-20 at 23:03, PBX wrote:
I have a quick question...
In the previous thread
I'm trying to come up with good jitterbuffer related settings for my
Asterisk boxes.
I ran 4 pings for about 2 days from my main Asterisk server to remote
Asterisk servers. During that time there were some large file uploads
which caused the max rtt to be quite large.
Here are the results:
Is anyone else having trouble receiving IAXTel calls? I don't know if
it's my config that's broken or IAXTel that broken. Several people have
given me their IAXTel numbers and calls to them all fail. I can call
FWD numbers via IAXTel just fine.
--Eric
--
Sample configs, scripts, more :
I managed to get inbound IAXtel working by setting it up the wrong way
(i.e. [iaxtel] as the last entry, etc).
You can call my IVR system at 700-923-3645. Extension 2101 is for
interactive services including talking clock, and callerid readback.
Extension 2102 is for system services like echo
I have the following in my extensions.conf:
exten = 21,1,NoOp(${CALLERIDNUM})
exten = 21,2,GotoIf($[${CALLERIDNUM} = ]?21|4:21|9)
exten = 21,4,Playback(/etc/asterisk/interactive-services/no-callerid)
exten = 21,5,Wait(1)
exten = 21,6,Playback(/etc/asterisk/interactive-services/no-callerid)
exten
Don't put the text in quotes. i.e. Festival(Would you like to play a
game?)
On Thu, 2003-10-23 at 18:07, Rich Adamson wrote:
I followed the script (closely) that you referenced (including the export),
and a ps ax shows:
21007 pts/2S 0:00 /bin/sh /usr/src/festival/bin/festival_server
That's not how it works. You specify the context of an extension in
zapata.conf, sip.conf, or whatever other channel.confs you use. The
context you specify there is for calls coming into the Asterisk server
from that channel or device. See my .sig for a URL for some simplified
example configs
The G729 codec needs a tty. The safe_asterisk script makes sure a tty
is available for Asterisk.
On Fri, 2003-10-24 at 16:11, Alejandro Ruiz wrote:
I hope somebody can help me out...
since I installed the G729 from digium, my asterisk box doesn't run in the
background any more.
I can only
Install the zlib and zlib devel packages
On Sat, 2003-10-25 at 12:18, [EMAIL PROTECTED] wrote:
Yes I do have the mysql and mysql-devel packages installed. With my very
limited knowledge of C/C++ here's what seems to be the culpret line right
before the error:
/usr/bin/ld: cannot
gzip is not zlib. On my Mandrake 9.2 system the zlib packages are:
zlib1-1.1.4-8mdk
zlib1-devel-1.1.4-8mdk
On Sat, 2003-10-25 at 12:36, [EMAIL PROTECTED] wrote:
Yes I do have gzip installed on that box. Any other ideas?
On Sat, 25 Oct 2003, WipeOut wrote:
[EMAIL PROTECTED] wrote:
make update, not make upgrade
On Sat, 2003-10-25 at 13:33, Rich Adamson wrote:
In attempting to make my asterisk server the latest and the greatest
tonight I attempted to upgrade my CVS. In the asterisk directory I ran
the make clean, executed the CVS update -d, and after all the
I've submitted http://bugs.digium.com/bug_view_page.php?bug_id=434
requesting that Digium put up a page with links with external Asterisk
related resources. If you have a web site with Asterisk related
information, patches, samples, documentation, etc, please add a bugnote
to the above URL.
Sounds to me like you are using inband DTMF, which won't work unless the
codec is ulaw or alaw. Use out of band DTMF aka rfc2833 or info.
On Mon, 2003-10-27 at 13:50, Steve Dolloff wrote:
I am seeing strange behavior that I don't understand. Voicemail2 and
voicemailmain2 work fine if I call
The mpg123 homepage is at http://www.mpg123.de/ Either follow the
instructions there for downloading and building mpg123 or use whatever
installation tool your Linux distro uses.
On Mon, 2003-10-27 at 14:25, [EMAIL PROTECTED]
wrote:
I would appreciate it if anyone can give me some instructions
w only works on Zap channels, as far as I know.
On Tue, 2003-10-28 at 17:42, David Broker wrote:
Hi,
I am using the manager interface to make a web-based interface to making
calls.
Our system is configured to dial a certain number to get an outside line,
you then hear the dial tone of
Humans tend to say Hello? (short burst of audio followed by silence),
and answering machines tend to say I'm sorry I'm not here right now,
please leave a message after the beep (long burst of audio followed by
a beep and silence).
So, basically you need to decide 1) what is audio and what is
There's a collection of external resources here:
http://bugs.digium.com/bug_view_page.php?bug_id=434
On Wed, 2003-10-29 at 05:49, Kris Edwards wrote:
Hi there,
I'm very new here and would like to know if anyone has reccomendations
on fundamental reading (other than the handbook) whick
If you have ANY chance of sending a fax over VoIP your codec MUST be
ulaw or alaw.
On Wed, 2003-10-29 at 03:52, Manuel Marín García wrote:
I have problems to send faxes using a fax machine connected to a ATA186 line
2. My sip.conf is
[1151]
type=friend
username=1151
secret=
A *bunch* of Asterisk resources, including AGI stuff is at
http://bugs.digium.com/bug_view_page.php?bug_id=434 If you know of
any resources not listed there, add them. I'm hoping Digium will put
these links on their web site under some kind of external or 3rd party
Asterisk resources page
See README.variables in the Asterisk source directory.
On Thu, 2003-10-30 at 10:13, James Coberly wrote:
Hi, after hammering out a message, due to several hours of fighting
format. I have it resolved.
Now, Is there a variable in Extensions that can be used as the incoming
callerID
You can only use inband dtmf if you are using the ulaw or alaw codecs.
On Thu, 2003-10-30 at 10:46, Shoval Tomer wrote:
Hi,
I'm having two problems.
First I'm using the xten x-lite program to communicate with
asterisk, and everything works fine except that DTMFs are not
transferred.
I want to have a list of companies providing services via IAX on my
Asterisk web page. If you know of a company that does this or run a
company that does this please e-mail me at [EMAIL PROTECTED] with 1) web
site, 2) contact info and 3) services provided. I don't want to put
pricing info on the
You are not using the new codec binary,
On Fri, 2003-10-31 at 12:59, Bartosz Jozwiak wrote:
Hello,
I have that problem with codec G729.
Please can somebody help me!
WARNING[16384]: File codec_g729b.c, Line 413 (load_module): Unable to
initialize va stuff: -1
== Detected 4
My problem with the Aastra phones is that NONE of them seem to have a
headset jack.
On Fri, 2003-10-31 at 14:23, Wade J. Weppler wrote:
Digium will sell you PowerTouch 480's along with the security code for
them, so you'll be able to use them with Asterisk.
-wade
-Original
If it doesn't find a host named ciscoccm1 then it will try to connect to
whatever host it got it's DHCP lease from. (assuming it's using DHCP, of
course)
On Sat, 2003-11-01 at 15:07, Brian West wrote:
Last I checked skinny firmware would try to connect to a host that would
resolve to CiscoCM1
There are links to several other Asterisk related sites at the bottom of
the page at http://www.fnords.org/~eric/asterisk/
On Sat, 2003-11-01 at 18:26, Brancaleoni Matteo wrote:
Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R
servers that do not support X windows under
You are missing a $ in front of {EXTEN:1}. It should be ${EXTEN:1}
On Sat, 2003-11-01 at 22:55, Matthew Enger wrote:
Hello,
I am trying to use 2 netjet cards under asterisk and isdn4linux. I am
having a hard time trying to get them to work in terms of dial out. Does
anyone have a working
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
As you can see they want a LOT of money. This is why I doubt there will
ever be G.723.1 codec available for Asterisk.
--Eric
On Mon, 2003-11-03 at
the DSP card.
On Mon, 2003-11-03 at 09:39, Gavin Hamill wrote:
On Mon, 2003-11-03 at 15:14, Eric Wieling wrote:
Licensing info for the G723.1 codec, direct from the holding company
that licenses the codec.
http://www.dspg.com/technology/LicensePricing.html
From what I remember when I
:41, Jeremy McNamara wrote:
Eric Wieling wrote:
The makers of hardphones prolly get their G72x licensing by using a DSP
that already has a license. The DSP can't be that expensive. I wish
someone would make a PCI card with something like 8 of these chips on it
and sell it cheap. Should
Dinesh Nair wrote:
On 16/11/2004 00:08 kido noagbodji said the following:
i have an easy way to install the codec under FreeBSD? It was tough
enough
to install asterisk even with the FreeBSD ports.
i do not believe that digium sells the g729 codecs for freebsd. however,
i too am a freebsd user,
Other than the standard codec issues? No. disallow=all and allow=ulaw
in [general] in sip.conf. NO other allow= lines.
Chris TenHarmsel wrote:
No one?
On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel [EMAIL PROTECTED]
wrote:
Hi all,
I've attached the output from asterisk with set verbose
Rgis MARTIN wrote:
Hi,
I know the question was already asked but I never found an answer to
this problem. So, I try again (things changes J)
Is there a way to play a specific message or sound from the start during
the dial command.
I want to do exactly the same thing that the m option of
Henry Devito wrote:
Hi, I have looked through the sounds files, but I think I am
overlooking the file that speaks Comidian Mail. I would like to change
this to a more friendly greeting, like Thank you for calling Comedian
Mail and instead of it saying Mailbox have it say enter your mailbox
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
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.
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Henry Devito wrote:
How do I implement the dial by name application on my IVR? I searched
through the archives with goggle and cant find the basic how to
information.
show application directory will show you.
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Huddleston, Robert wrote:
Is directory included w/ Asterisk or external app... I'm running older
release * so j/c
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
Angel Diaz wrote:
Hi all,
Does somebody know what's new with SS7 and * ?
I'm very interested. Is it ready ? I'm prepared to pay if necessary.
Join the asterisk-ss7 mailing list at http://lists.digium.com/
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Joe Greco wrote:
Who to generate ring tone to a calling party when the call is passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
Did you tell Asterisk to indicate ringing?
Asterisk will ALWAYS indicate
Andrew Kohlsmith wrote:
On November 17, 2004 04:30 pm, Eric Wieling wrote:
Asterisk will ALWAYS indicate ringing if it can.
The r option to Dial, Playtones, and Ringing are all hacks/workarounds
for when Asterisk cannot indicate ringing to the calling party. You
should diagnose and fix the real
Joseph wrote:
On Wed, 2004-11-17 at 14:58 -0700, Joseph wrote:
[snip]
Asterisk will ALWAYS indicate ringing if it can.
The r option to Dial, Playtones, and Ringing are all hacks/workarounds
for when Asterisk cannot indicate ringing to the calling party. You
should diagnose and fix the real
Joe Greco wrote:
I have to confess that I don't understand the real problem, then.
What else are you supposed to do when you've already answered a channel
and you then want the caller to hear ringing? It didn't seem to work
automatically, and I'm certainly not convinced that it should.
That
Kyle Hagan wrote:
Alot of people have needed the SIP firmware for the Cisco 79xx phone. I
found a link for them..
http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html
Well, yeah, I guess stealing it is one way.
[EMAIL PROTECTED] wrote:
I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
I am using the Wildcard X100p.
Part of that delay is just waiting for the X100P to dial. Part of that
delay may be overlapping dialplan entries.
exten =
[EMAIL PROTECTED] wrote:
[outgoing]
ignorepat = 9
exten = _9,1,Dial(Zap/1)
Dial(Zap/1/)
Note the extra / for the FXO port.
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Giovanni Powell wrote:
Or can i set the Interval of ingorepat
ignorepat is processed without any delay.
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Sean Kennedy wrote:
Steven Critchfield wrote:
To the mailing list admins,
The idiots running the pliva.hr mail server are not responding to
messages about their broken mail server and it's insistence to send
everyone here a copy of a mail saying The recipient: MARIO SPOLJAR is
no longer PLIVAs
Joe Greco wrote:
Anyways, Not That I Would Encourage Anyone To Do This, but NFR's of Netgear
products are available at half off list ($875 for the switch in question)
to Powershift partners. That's gotta be one of the better prices for that
switch at this time.
Dell has some 48 port supposedly
Brian C. Fertig wrote:
I have a 3348 they don't do PoE. They do QoS and do it well. I don't
know about the upper models..
I stand corrected.
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Martin List-Petersen wrote:
You can't, the T100P is a unchannelized T1 card.
This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1)
If you want to use it with HylaFax you need either SpanDSP OR an analog
port on Asterisk in addition to the T100P.
A search of the mailing lists
Michael Devenijn wrote:
We are located in Belgium and just ordered a PRA line, the telco asked the following questions :
- 120 or 75 ohm ?
- Support for CRC4 yes/no ?
- B channels in 2 way ?
1) Neither. Digium cards require an RJ-45 connection. Search the
mailing list for info on this. I
Google: Results 1 - 10 of about 149 from lists.digium.com for Unknown
error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html
http://lists.digium.com/pipermail/asterisk-users/2003-November/028105.html
Stefano Finetti wrote:
From: Eric Wieling [EMAIL PROTECTED]
Google: Results 1 - 10 of about 149 from lists.digium.com for
Unknown error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.html
http://lists.digium.com/pipermail/asterisk-users/2003-November
Date: Fri, 19 Nov 2004 08:10:32 -0600
Stefano Finetti wrote:
From: Eric Wieling [EMAIL PROTECTED]
Google: Results 1 - 10 of about 149 from lists.digium.com for
Unknown error 500.
Specifically:
http://lists.digium.com/pipermail/asterisk-users/2004-April/042912.ht
ml
http://lists.digium.com
Martin List-Petersen wrote:
Citat Eric Wieling [EMAIL PROTECTED]:
Martin List-Petersen wrote:
You can't, the T100P is a unchannelized T1 card.
This is 100% wrong. The T100P supports Channelized Voice T-1 (aka CT1)
If you want to use it with HylaFax you need either SpanDSP OR an analog
port
Eric Hall wrote:
Here is what I was trying to do
Telco gives me a PRI It connects to my Asterisk via Wildcard T100P using
a 2nd Wildcard T100P I would like to connect with a X-Cable my HylaFax
server.
My question is will a Wildcard T100P work in a Hylafax server?
No it will not.
Your only option
Altus Snyman wrote:
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example:
exten = s,5,Dial(SIP/302,25)
exten = s,6,Hangup
exten = s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still
Peter Landy wrote:
Yes I did. Does anyone have a working list of libraries and versions. I have
tried with different releases of H323 and they all give different errors.
Also is it necessary to compile the H323 under asterisk src/channels/H323
as this also bails on errors. The rest of my asterisk
Shaun Tierney wrote:
From what I've been reading about the callprogress option, it seems like it
will work properly only with a T1 or PRI in the US. Is that correct or are
there still issues with call progress detection even if those qualifications
are met?
If you ask me it doesnt' work well
Michael Welter wrote:
Using the |caller parameter, TxFax injects the fax tone (CNG) onto the
line. With the CNG tone, asterisk is unable to detect the busy tones.
If I were to remove |caller then the receiving station wouldn't
receive the CNG tone and possibly not direct the call to the fax
Matthew Boehm wrote:
-- Executing Dial(SIP/3044-8d49, ZAP/g1/15124512424|60|t) in new
stack
Nov 22 10:59:32 WARNING[1126867776]: chan_zap.c:4159 zt_write: Cannot handle
frames in 256 format
show codecs in the Asterisk CLI will tell you the number of each codec.
If you want to use G729 and the
Matthew Boehm wrote:
Well, it seems that Zap cannot do 729 at all:
channels/chan_zap.c (line 4156):
if ((frame-subclass != AST_FORMAT_SLINEAR)
(frame-subclass != AST_FORMAT_ULAW)
(frame-subclass != AST_FORMAT_ALAW)) {
ast_log(LOG_WARNING, Cannot handle frames in %d
Shaun Tierney wrote:
Ok, so if I turn callprogress off, and try to connect a call which is
bridged between an incoming line and an outgoing line, will it treat the
call as being answered once it is bridged or once it is actually answered on
the outgoing T1 trunk?
No, it will not be answered unless
Cristian Manoni wrote:
Is Directed Call Pickup supported in asterisk?
(http://voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup)
No.
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To
WipeOut wrote:
The problem is that I don't think Asterisk is causing the problem (not
entirely anyway), I think its the internet and that IAX is too sensitive
to packet loss so when the packet loss exceeds a certain threshold it
just drops the call instead of trying to recover and maybe having
Bastian Schern wrote:
Does nobody know whether this is possible or not?
Bastian Schern schrieb:
Hi to everybody,
is it possible to use ISDN Call Deflection with a ZapHFC card?
Call Deflection example can be found at
http://www.junghanns.net/asterisk/page9.html This looks like it's CAPI
Carsten Bock wrote:
Hi,
el Flynn wrote:
Carsten Bock wrote:
Hi there,
How do i setup asterisk, so that in the CDR's is only the time, which
the line actually was connected? Not the time, the line was up, but
the time the user was able to talk to another user.
isn't it already in there?
I
Damian Minkov wrote:
How can i control the codec for the calls. For example I have 3 SIP
phones registered to asterisk
The firs two are in the local area network (behind nat)- I want to use
g711 between them and to connect directly (canreinvite=yes)
and the third is in internet - want all calls
kido noagbodji wrote:
Hello all,
I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering
if by any luck, i could make some magic happen with asterisk ... If asterisk
does not support it, is there any PSTN to H323 or PSTN to SIP gateway that
support this dialogic card and
Leonardo J. Tramontina wrote:
I don't think the problem is in the number of digits because, as I said,
I'm not making a real call... I'm not redirecting to the real number
482343400 (my Asterisk is not connected to another network)... I just
want to allocate one channel to test my TE110P!!!
Do
Leonardo J. Tramontina wrote:
No, I don't have anything connected on the TE110P.
After the Unable to create channel of type 'Zap' (cause 0) message, I
also get the CHANUNAVAIL...
Is not possible test a channel from the card without connections on it??
Correct.
,Answer
exten = 3300,2,Ringing
exten = 3300,3,Wait(1)
exten = 3300,4,Playback(/etc/asterisk/sounds/pls-wait-connect-call)
exten = 3300,5,Wait(1)
exten = 3300,6,Ringing
exten = 3300,7,Wait(1)
exten = 3300,8,Dial(Local/[EMAIL PROTECTED]Zap/25,,r)
exten = 3300,9,Hangup
--
Eric Wieling
of these patches are in CVS versions after March 18 2004.
http://www.fnords.org/~eric/asterisk/
--Eric
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
On Tue, 2004-03-23 at 09:49, Eric Wieling wrote:
I'm having a similar problem with 0.7.2 but ONLY if I dial multiple
destinations at the same time. Here is a copy of my extension section
that does NOT provide ringback no matter what I do. In this example the
caller hears ringing while
Express Station
7960 IP Phone with one Station User License
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss
licenses the
same cost?
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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to the hangup.agi script.
I have tried
$var = $ARGV[0];
$var = $ARGV[1];
but still can not get the passing variable value.
--
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In a related story, the IRS has recently ruled that the cost of Windows
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.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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is going on with the SIP and RTP packets.
Yes, that's the next thing I have to try. Hopefully this evening.
Interested to see you are just up the road: I'm in Winchester.
reinvite= is a myth. It does not exist.
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related
,Dial(Zap/1,30)
exten = 2111+vmbe+true,2,VoiceMail(u2196)
exten = 2111+vmbe+true,102,VoiceMail(b2196)
# Mailbox 2111 in voicemail context [default] does NOT exist
# Ring forever
exten = 2111+vmbe+false,1,Dial(Zap/1)
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Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related
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can output:
faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4
tifflzw tiffpack
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Michael Welter wrote:
I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM
3CNJPSE power injector. Can I put one of these behind my LAN hub and
power all the phones, or do I need one for each phone?
From the spec, it looks like PoE tries to discover whether a device is
powered
Jeb Campbell wrote:
Anyway, the only stuff off list was trying to debug the connection.
1. With a crossover there is no sync (YELLOW and RED alarms)
2. With standard cable I get a pri error that they think they are the
NET, but we are the NET.
(This is asterisk 1.0 stable and the directions from
?
Cheers
Peter
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In a related story, the IRS has recently ruled that the cost of Windows
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. I believe that the CVS stable as of a few
days ago fixes the problem. If not, put a r option at the end of the
dial like to work around the problem. e.g. Dial(Zap/1,20,r) If you
don't use a timeout then use something like Dial(Zap/1,,r)
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Eric Wieling * BTEL Consulting * 504-899-1387
On Wed, 2004-03-31 at 20:57, Gene Kochanowsky wrote:
Thanks Eric. Is this the CVS branch you are referring to? -
# cvs checkout -r v1-0_stable asterisk
Yes.
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