Is there a way to tell asterisk to start recording DURING a call?
Can I flash, then *XX, and then flash back to my call while asterisk
records?
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FC4 - a good choice.
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
Date: Tue, 03 Jan 2006 09:57:17 -0800
From: Mike Fedyk <[EMAIL PROTECTED]>
S
I have an asterisk
1.2 (beta 1) install trying to use the spool feature but the call always
fails. I confirmed a manually dialed call on the same trunk works
fine. The error always ("reason 8") appears to be
AST_CONTROL_CONGESTION. But this is not reasonable..here is the
log:
-- At
Many of you current
cid_rewrite (v1.0.0) users probably noticed that your 411 lookup is broken,
thanks to another change by the 411.com folks. So we fixed it :) The
latest changes include:
1. Adapt to new
411.com format
2. Improved address
conversion and extraction from reverse lookup (re
Consider using fax2mail (available at www.generationd.com). It's a free
add-on to asterisk to email you incoming faxes in PDF/TIFF format. Nice and
simple.!
Also available is mail2fax from same source.
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;.
Not a big deal, but a nice start! Does anyone else of
updates/recommendations I should roll in before I release this script?
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your busine
E .
T: (519) 672-8238
E: [EMAIL
On a recent install
of ast 1.2 (b1) I noticed something strange in the CDR records (in mysql).
The caller ID name and number contained extra quotes for calls outbound
(inbound was fine).
Below is an example
of the extensions.conf excerpt, and an excerpt from my sql. Can anyone
explain
send to VM if 2.
It's not that hard I just don't want to reinvent the wheel.
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
__
tag in the tiff. (This is easier than
rotating the image bits).
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original Message-
From: Andrew Nowrot [mailto:[EMAIL
Are you sure you want to resize the image? Do you
perhaps want to rotate it?
Do you want to rotate
every image before it arrives? (At 200 dpi, it seems that you are
just switching landscape to portrait)
Michelle
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work
I'm not sure I
follow. As I understand it, app_rxfax and app_txfax are designed to
transfer fax data, over any asterisk channel. Since the fax is converted
to audio over the channel, and the channels are IP, isn't this fax over
VOIP?
I am successfully
using app_rxfax to receive (analog)
Trying using the
fax2mail script from www.generationd.com
You will at least
get a log showing call received, conversion to tiff/pdf, mailout,
etc.
Michelle
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your
business...
T: (519) 672-8238E:
[EMAIL PR
Ooops - my mistake. But hoodahek is getting CID names that I never saw
beforeso where is it getting those?
Michelle Dupuis
Technical Support Specialist
Oxford Consulting Group Ltd.
Making IT work for your business...
T: (519) 672-8238
E: [EMAIL PROTECTED]
W: www.ocg.ca
-Original
Instead of
anywho.com, try hoodahek. This is a great addin for Asterisk - it uses the
411.com service to resolve names.
(it's also
compatible with mail2fax in terms of naming)
Michelle
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your
business...
Thanks for the great
feedback! We now have an updated fax2mail (version 2.0) for download (at
www.generationd.com). Fixes
include:
1. Improved
detection of the number of fax pages
2. Handling of
"!" character in the name (for those users of the "hoodaheck" module)
3.
Correction of
The fax2mail and
mail2fax scripts can be found on www.generationd.com
Michelle
DupuisTechnical Support SpecialistOxford Consulting Group Ltd.Making IT work for your
business...
T: (519) 672-8238E:
[EMAIL PROTECTED]W:
www.ocg.ca
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I have app_rxfax and
app_txfax loading properly (they are in my modules.conf and they show as loading
without error when I start asterisk)
I create .call files
(in the outgoing directory) and associated tiff files but app_txfax never acts
on them! They just sit there forever. I checked t
I need to have a
bash script trigger Asterisk to goto a particular extension (as if that
extension were dialed). I can't find any documentation for
this.
Does anyone know of
a way for a bash script to Dial an asterisk extension? (I'm trying to make
a bash file call the app_rxfax applicat
Thanks for all the
feedback! I have posted the latest fax2mail and mail2fax scripts on the
site www.generationd.com
For those new to the
scripts, they are a friendly interface to asterisk to make it easy to
send/receive faxes by email. They now include email confirmations of
fax submis
Roman:
I created two bash
scripts called Mail2Fax and Fax2Mail for use with the asterisk
sever.
They leverage the
app_txfax and app_rxfax scripts, along with ast_fax. They make using these
apps a lot easier, including being able to mail to [EMAIL PROTECTED] for outgoing faxes and then
Has anyone configured ast_fax (sending faxes via asterisk)
with sendmail? The creation of rules to trap all numbers [EMAIL PROTECTED] seems too
complicated. Does anyone have setup details to share? (I don't want
to switch MTA's).
As a workaround, I could
launch the app automatically fr
I found the
problem! Installing spandsp .3 created a symlink that was not
removed. Installing spandsp .2 did not replace the link. That cause
the wrong library linking in
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I managed to compile
app_rxfax and app_txfax against the latest asterisk (1.2 beta 1). When
trying to load the app_rxfax module I get this error:
[app_rxfax.so]Oct 4 12:52:25
WARNING[3701]: loader.c:314 __load_resource:
/usr/lib/asterisk/modules/app_rxfax.so: undefined symbol:
fax_set_
I'm trying to get
ast_rxfax and ast_txfax compiling with Asterisk 1.2.1 beta. The two
ast_?xfax files don't compile:
gcc -pipe
-Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
-march=i686
-fomit-
I'm looking at
switching VOIP providers, but want to ensure I move to a company with sufficient
capacity.
Can any Canadian
VOIP users post/email me with feedback on their providers?
I'll post the
results for all to read..
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I'm running asterisk
1.2b1 and all seems to be working right in general. I load modules
explicitly in modules.conf, and since my upgrade ast 1.09 I have only one
problem:
The LEN function
(length of string). What module do I need to load to get this string
handling function?
Thanks
Can anyone recommend
a good IAX provider offering numbers in Toronto and
Detroit?
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Has anyone tried HooDaHek with asterisk 1.2b1 ?
I know the data structures have changed somewhat...
MD
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I wish to completely
remove all signs of asterisk from a Linux machine. Is there an automated
uninstall (eg: make uninstall)?
If not, does someone
have a link to a page describing how to uninstall asterisk (v
1.0.9)
Thanks,
Michelle
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riginal Message-
From: Technical Support [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 06, 2005 10:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion';
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk overheating on VIA Epia
MSeriesmotherboard
You can dramatic
You can dramatically reduce the heat from your EPIA board by turning on CPU
scaling! Once we turned it on, the heatsink was cool to the touch. (Even
with asterisk running).
MD
-Original Message-
From: Angus Comber [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 06, 2005 8:16 AM
To:
annel. If you specified a parameter
(like callretrun=yes or callprogress=yes) that the LEC (Carrier) didn't
like, it would not accept the call. If group 5 works correctly for
outbound calls, I would model group 3's defninitions after group 5.
Joe
"Mark Messmore, Technical S
ng sure you've got all of the channels assigned to
groups. My zapata.conf is much simpler:
signalling=pri_net
group=1
channel => 1-23
When it dials, then you will see the calls going out on Zap/1-1 or
Zap/1-2, etc.
Good luck; and have fun!
Joe
"Mark Messmore, Technical S
Title: Message
I've posted this problem a couple of times before with
little or no response. Basically I have a T100P in my * box.
Incoming calls are working great. However outgoing calls are not working
at all. I've copied a previous post into this message which should have
all the nec
K...maybe this was stated earlier in the conversation...but what's the
deal with the phone? Or was this phone just being carried around by
everyone and ripped apart?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, April 05,
Luciano,
I was having the same thing happen after updating to that code...but
since mine is in production I had to quickly go back to the code from
two weeks ago. I know it's not a "solution"...but if you really need it
back up now you might want to do that.
Mark
-Original Message-
Fr
CTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jessie Bryan
Sent: Friday, March 26, 2004 2:43 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] T1 outgoing calls problem.
Mark Messmore, Technical Support, University Telcom Inc. wrote:
> -- Executing Dial("SIP/mark-2d08", &quo
Title: Message
OK...I've got an *
box with a T100P in it. For the most part incoming calls are going through
just fine. Outgoing calls, however, I'm having some more trouble
with. Whenever I make an outgoing call, the call begins, however after the
dialing process all I hear is dead air.
Title: Message
just
having a post on the list would work...a lot of times searching through list
archives brings a lot of good info...another good place is on the
wiki.
Mark
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umar
SearSent
lf Of Mark
Messmore, Technical Support, University Telcom Inc.Sent: Monday,
March 22, 2004 1:15 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
T100P not ringing.
Thanks for the response. Here are two contexts from my
extensions.conf. The number being dialed is in the
7;t have an "r" in your dial command. That
lets the calling party hear a ring.
e.g.
Dial(SIP/1234|20|Tr)
-sb
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Mark
Messmore, Technical Support, Univer
Title: Message
I posted this
problem another time, but with another problem tied in...so let's try this
out.
I've got an X100P
and a T100P on the same box (the x100p was initially for testing, but since it's
working fine we are still using it). However, the X100P is tied into a
differen
Hey all. We've just recently purchased a T100P in order to provide VoIP
to a remote office. We've interfaced it with a DS1-formatter on our
Mitel GX5000 switch. I realize that plugging the * PBX into this class
5 switch isn't the best situation to have in the world...but hey it's
what we've got.
Just as a matter of curiosity...how often do most of you update your *
installation from the CVS?
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Is anyone presently using the Sipura SPA 2000 for faxing? I was about
to look into it and just figured that I would ask to see if anyone ran
into any snags, problems, etc. Thanks.
Mark
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out. If I hear anything I'll be sure to let you know.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad
Jordanovic
Sent: Tuesday, March 02, 2004 3:58 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sipura 2000 not ringing.
Mark Messmor
ringing.
Did you try to upgrade the firmware?, some issues we saw with rtp
stream, went away after a firmware upgrade.
http://www.sipura.com
-SamW
-Original Message-
From: Mark Messmore, Technical Support, University Telcom Inc.
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 02
I was just wondering if anyone has had this situation...or one similar
to it.
I've got a Sipura SPA 2000. After hooking it up and configuring it with
my * box, it has worked well. From both lines we are able to dial out
at any point in time. However after a few minutes (5-10 usually) the
Sipu
way...it's alright now.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: Tuesday, March 02, 2004 10:08 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Incoming calls.
OK...first off than
OK...first off thanks for the responses. However, I'm still having the
same sort of issue. I've looked through the two places that you
gentlemen suggested, and am still having the problem. Here is the error
message that I am receiving:
"-- Starting simple switch on 'Zap/1-1'
Mar 2 09:58:18 WAR
Hey...I'm sure that this is something very simple that I'm missing (I've
had * just about 2 weeks or so now and just finally got my x100p card
in). Here's my simple setup.
VoIP Network--*--PSTN via X100P
I can make outbound calls from my VoIP network just fine. However, when
trying to dial in
Oh...and here is the error message from my error_log...fyi.
[Tue Feb 24 11:52:59 20 04] qview.pl: Invalid argument at
/var/www/cgi-bin/qview.pl line 23.
Mark
Has anyone ever had the following error when using qview.pl?
"Invalid argument at /var/www/cgi-bin/qview.pl line 23."
Just wond
Has anyone ever had the following error when using qview.pl?
"Invalid argument at /var/www/cgi-bin/qview.pl line 23."
Just wondering.
Thanks in Advance.
Mark
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Have you tried using a soft-phone just to see if it's the phone or
something with your * system?
Also, have you tried doing a asterisk -vgcr to see what exactly
is happening at this disconnect?
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
I've looked through a lot of different pieces of documentation regarding
*'s ACD functionality. Is there any one place in particular with a good
amount of documentation on it?
Thanks
Mark
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I had this with X-lite (on windows though) where I could hear it on one
end but not on the other. On the end where I couldn't hear audio I did
this
Advanced System Settings --> Audio Settings --> Silence Settings -->
Transmit Silence --> Change this to "Yes"
That worked for us. Give that a sho
onf file.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Messmore, Technical Support, University Telcom Inc.
Sent: 18 February 2004 19:57
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] softphone configs?
k...Here you go. (that is if attachme
EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] softphone configs?
What does your sip.conf look like? Please include it in your next
message in
its entirety.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Messmore,
Technical Support, University T
I checked them out and they looked fine...still got the error. I
removed them out of the sip.conf entirely...still got the
errorsigh...
Thanks for the response though.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Girish
Gopinath
Sent: Wednes
I've tried using the x-lite softphone as well as sjphone. I've gone
over my configurations a dozen times...and I always seem to get the
following error:
Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request:
Registration from 'Mark ' failed for
'192.168.5.64'
FYI...I'm trying to do
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