Jeremy Gault wrote:
We've had the same problem here ever since we upgraded to CVS-HEAD.
When someone placed a call to a number that was busy, they would just
receive the call cannot be completed recording we have setup at n+1.
Someone please correct me if I am wrong. *dons asbestos armor,
On 10/14/2005, Jeremy Gault [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
Ever since this upgrade, the system is jumping n+101 if it gets a busy
on a Dial command, it is now proceeding to the next priority (n+1)
Has something
The host you're registering with sets the expiry parameter to 60 seconds in the reply message. Use sip debug to see SIP messages running.
On Fri, 2005-10-14 at 17:00 -0300, Ricardo Poppi wrote:
Hi list!
Im connecting a Brasilian voip (- gvt.com.br -) provider through my
asterisk box and
Hi!
I would recommend a 8
port fxs MOSA 3708 from Vodtel. Works perfect with Asterisk and a
reasonable priced compared to Quintum Tenor. Can be bought on www.bobascom.com. The webshop is in Europe but sell equipment to customers all over the world
Anders
From:
[EMAIL
Where can I change the Duration length of an INFO
packet?
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=5
Duration=250
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Chris Johnson, Network Operations Manager
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On Fri, 2005-10-14 at 17:10 -0400, Chris Johnson wrote:
Chris Johnson, Network Operations Manager
NCisp.Net
919 736 9775 / 919 553 7160
[EMAIL PROTECTED]
This problem seems to have been fixed (at least for me) in the latest CVS-HEAD I snagged (13 OCT 2005).On 9/23/05, Trey Blancher
[EMAIL PROTECTED] wrote:
I'm trying to get ChansSpy to work. It works, in
the pass/fail sense, but it is difficult to understand the various
speakers. I can hear users
On 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
I recently upgraded my Asterisk system to the latest CVS-HEAD
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-12 13:34:09 UTC
Ever since this upgrade, the system is jumping n+101 if it gets a
busy
Message: 13Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT)From:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk/Cisco Call Manager 3.3To: asterisk-users@lists.digium.comMessage-ID:
[EMAIL PROTECTED]Content-Type: text/plain; charset=iso-8859-1I need to pick all the Asterisk and Cisco People a
Seems to, mostly, but not all the time.. There have been some
previous posts about this phone. I'm waiting for the next firmware
release before testing further.
Mute kills audio in both directions. Mine doesn't seem to want to
dial any numbers that have * in them like *98.
My Sipura 3000
I would have to agree - your easiest route is to upgrade to CCM 4.0+
with SIP trunk support..
On Fri, 2005-10-14 at 16:55 -0500, Paul Davidson wrote:
Message: 13
Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT)
From: [EMAIL PROTECTED]
Subject:
I'll throw in a few requests as well-
A pause feature.
The ability to mark a recording urgent.
The ability to change the prompt features around and edit voicemail prompts,
recording abilities, while retaining defaults and customizations for
different extensions.
I'm still studying AGI's
Not sure what asterisk version you're using, but in cvs head you might
try the new method:
Set(CALLERID(name)=4989427)
Set(CALLERID(number)=4989427)
Mojo
Doug Lytle wrote:
René Enskat [Teamware GmbH] wrote:
My number is not submitted.
I updated my asterisk but this error still
Should work that way. Right now it's coded so that if the person being called doesn't acknowledge with DTMF, it'll timeout and go on to the next caller.
If it doesn't reach anyone in the whole list, it then exits just as the Dial application does with DIALSTATUS populated and it's then your
CF -
You're right. Most of this can be done with the dial plan. I wrote the app though because I wanted to be able to have the option to work with both channels at the same time through a threaded model. The dial plan doesn't let me do that, and there's no reason it should.
So, in this
can one use a milliwatt test line from a telco in a different
areacode/prefix? does the long-distance transmission destroy levels
unpredictably?
Shaw Terwilliger wrote:
On Fri, Oct 14, 2005 at 11:38:00AM -0600, Rich Adamson wrote:
A milliwatt generator creates an audio signal at 1,004 hz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mojo with
Horan Company, LLC
Sent: Friday, October 14, 2005 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Calibrating both RX and TX gain?
can one
Hi Andrea,
How do you start a weekly-rotation log?
Do we need to do it manually through CLI? or can we set it somewhere?
Thanks.
AK
On 10/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thank you very muchI decided not lo lower the log information (leaving all: full
Hello all,
Okay when you are done laughing at the simplicity of this question could
someone show me please what I have wrong in the following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a single digit from 1 to 9
inclusive goto 15, if it is a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
William M.
Sandiford
Sent: Friday, October 14, 2005 2:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Busy not jumping n + 101 anymore
On 10/14/2005, William M. Sandiford
Same reason why some ISP's charge so much for IP addresses. Because they
can.
Every carrier ISP I have ever looked at, Unlimited IP Addresses, but when
a company chooses that particular carrier, they charge between $5 and $20,
seen as high as $25, per IP address to their clients.
Why? Once
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9] ]?15:3);
What this is supposed to do is if numdial is not a
Hello All,
Just a question, I have an adit600 and I am looking for a way to pull
the incoming cid into asterisk.
Does anyone know if this is just not possible via t1? Or is it only
available on PRI?
Thanks,
Greg
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New asterisk user,
pretty much set up except for spandsp. I get the following when trying to
compile:
app_rxfax.c app_rxfax.c: In function
`phase_e_handler': app_rxfax.c:92: error: structure has no
member named `cid' app_rxfax.c:92: error: structure has no
member named `cid'
Oddly enough, I believe it's mentioned in UPGRADE.txt.
Kris Boutilier wrote:
n 10/14/2005, William M. Sandiford [EMAIL PROTECTED]
wrote:
I recently upgraded my Asterisk system to the latest CVS-HEAD
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-10-12
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in the
following statement?
GoToIf($[${numdial} != [1-9]
Administrator wrote:
New asterisk user, pretty much set up except for spandsp. I get the
following when trying to compile:
app_rxfax.c
app_rxfax.c: In function `phase_e_handler':
app_rxfax.c:92: error: structure has no member named `cid'
app_rxfax.c:92: error: structure has
On Friday 14 October 2005 20:50, Eric ManxPower Wieling wrote:
Oddly enough, I believe it's mentioned in UPGRADE.txt.
Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt.
-A.
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On Friday October 14 2005 8:57 pm, John Millican wrote:
On Friday October 14 2005 8:26 pm, Samy Antoun wrote:
--- John Millican [EMAIL PROTECTED] wrote:
Hello all,
Okay when you are done laughing at the simplicity of
this question could
someone show me please what I have wrong in
Yes adding the + before ${EXTEN} to my extensions.conf did the trick
exten = _1800NXX,1,EnumLookup(+${EXTEN})
Thanks.
Apparently ENUM now REQUIRES a + at the beginning of the number to query.
EnumLookup(+18886532145)
No I didn/t see it documented anywhere.. It seems to require it even
On Wed, Oct 12, 2005 at 12:01:17PM +0200, gincantalupo wrote:
2) implement a log rotation or similar of the full log ?
I see the full log grows a lot (about 100 MB per Month)
Use logrotate.
logrotate (which comes with most linux distributions) will indeed save
you the dirty work of
zdummy NEEDS the usb drivers to load the clock,
if you disabled usb from bios, then the kernel is not loading usbci
On 10/13/05, Bruce Ferrell [EMAIL PROTECTED] wrote:
Hi all,
Trying to build ztdummy on an old redhat 7.3 box running kernel
2.4.20-43.7.legacysmp. Yes, I have the kernel
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