Wow. How on EARTH do these people stay in business? Just running the
law of averages and hoping it works out?
$10 a month for unlimited routing through their 800 number seems like
a risky gamble for them.
On Nov 12, 2007, at 9:21 PM, Paul Hales wrote:
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn’t find much up there.
Thanks
Softkeys running both SCCP and SIP firmware are
Hi All,
Interfaces of my PBX are as follows,
Toshiba dk28
CO Lines (to telcos) : 12 - (2 free)
Digital extensions : 8 - (full)
Analog extensions :18 - (full)
Toshiba dk280
CO Lines (to telcos) : 8 - (1 free)
Digital extensions : 16 - (5 Free)
Analog extensions : 16 - (1 free)
Toshiba
Hello,
I currently have a pretty standard 1.2.21 Asterisk system running purely
SIP termination (no zap/IAX/H323..etc).
We have an auto-dialing system that generates calls via the manager API.
The system runs beautifully until it gets to about 200 calls. I can
generate these calls in quite
Hello,
While trying to install H323 support for asterisk, I missed one step.
After compiling files in channel/h323, need to select chanh323 from the menu
and compile and install asterisk.
cd asterisk
* ./configure
* make menuconfig
channel drivers-chanh323
save the setting by giving
Nick Adams wrote:
The kicker: I can't seem to get past this 200 call point even though the
What does your console show at this time?
When testing, I've noted the 200 call limit was because I had too many
open files. I had to increase this by typing ulimit -n 4096 before
starting
Citel SIP handset gateways at www.dtasia.net
Rupert Utteridge
Director - Sales Marketing
Digital Techniques (Asia) Limited
4 The Lee
Middle Cove, NSW, 2068
Australia
Tel: +61 2 9037 4191
Mobile: +61 424 373 516
Web: www.dtasia.net
Message: 1
Date: Tue, 13 Nov 2007 23:00:54 -0500
From:
Hi
I have a customer who is using Linksys 942 phones.
When they try to transfer a call the Asterisk CLI
reports that both legs of the call must exist on the
server. The call they are trying to transfer then
drops.
Does anyone know why this is and how to fix it?
TIA
Regards
Jon
Hi,
Any reason why I can not get the php exec() function to execute a shell
command inside an agi script?
Thanks.
Andre
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On Nov 14, 2007 12:52 AM, Erik Anderson wrote:
On Nov 13, 2007 11:44 PM, Mohammad Shokuie wrote:
HI Erik,
thanks for your post, Actually im sending new posts not
replying but if you see them correct, how come its wrongly
viewed for me. Are you using a speciall software to view
The Cisco Documentation states that you can modify standard and
nonstandard softkey templates. They may not be xml files. I just
assumed they were xml since that is what is used to configure the phone.
Here is snip from the 7911G documentation that states you can configure
the private key (which
I have asterisk 1.2.18 running on a new system we just installed. Although
I've used AGIs many times in the past, I'm stumped on this one. It may just
be a simple issue that I need another eyeset to look at.
My AGI does the following:
#!/usr/bin/perl
#Load a few modules...
use Asterisk::AGI;
On Wed, 14 Nov 2007, Matt wrote:
I have asterisk 1.2.18 running on a new system we just installed.
Although I've used AGIs many times in the past, I'm stumped on this one.
All seems fine. If I run the script from the command line it works as
expected:
However, when actually running in
I looks like this gateway is the FXS - it allows Meridian handsets to talk
to a SIP pbx.
I need a an FXO gateway - which allows Meridian PBX digital lines (not
trunks) to talk to a PC/SIP/etc.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi,
I have an older phone with touch screen from Philips. It have ti connected
to Sipura 3000 FXS port and majority of features work ok.
But phone also has touchscreen and web browser that I'd love to use for
accessing my local web pages. But the phone only allows me to setup ISP
phone number
Quoting Michelle Dupuis [EMAIL PROTECTED]:
there were the meridian ATA boxes that plug into a digital extension
port and provide a standard analog extension - you would probably be
better off with several of those - that is what they were actually
intended for (analog fax, voicemail
-rwxrw-r-- 1 asterisk asterisk 1053 Nov 14 08:54 GetEmailFromDID.agi
asterisk 3348 0.0 0.2 27024 9380 ?Sl Oct01 0:01
/usr/sbin/asterisk -U asterisk -G asterisk -v -g -p -U asterisk -G asterisk
On Nov 14, 2007 9:17 AM, Brett Crapser [EMAIL PROTECTED] wrote:
On Wed, 14 Nov
No.
AGI script is just like any other PHP script, the only difference is
that STDOUT/IN are connected with the Asterisk thread that launched
the script.
You should run the script directly and see what the problem is.
- Moy
On Nov 14, 2007 6:51 AM, Andre Courchesne [EMAIL PROTECTED] wrote:
Hi,
We are using 2 different incoming trunks.
The first one is alsion.com and is sending INVITE with phone number in
the INVITE line whereas plugandtel put the callee number only inside the
To: Section.
Marco Mouta a écrit :
Could you describe in detail how did you fall into this situation, I mean
Does anyone have any solid documented evidence for best practices for IVR
Trees? I'm just curious. I did some google searching this morning, but
only found one article from TMC.What I'm looking for are studies
showing, if a customer must go to an IVR, what they prefer, over what they
prefer
what does agi debug says?
what if you run the script from the command line and you fake the
asterisk input?
Regards,
On Nov 14, 2007 8:33 AM, Matt [EMAIL PROTECTED] wrote:
-rwxrw-r-- 1 asterisk asterisk 1053 Nov 14 08:54 GetEmailFromDID.agi
asterisk 3348 0.0 0.2 27024 9380 ?Sl
Is the Whats New at Digium the Asterisk Company message I got from
[EMAIL PROTECTED] really from Digium?
If so I suggest to send it from digium.com and not to use those
shady Eloqua redirect URLs.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Make sure /usr/bin/perl can be reached.
Also try in your CLI:
agi debug
Same case happens when I do not have php-cli installed for php AGI scripts.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
From: [EMAIL PROTECTED]
[mailto:[EMAIL
[EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v
This is perl, v5.8.5 built for i386-linux-thread-multi
Debug shows nothing:
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
AGI Tx agi_request: GetEmailfromDID.agi
AGI Tx agi_channel: Zap/23-1
AGI Tx agi_language: en
AGI
I received this message as well. I considered it spam, as I¹ve not
voluntarily signed up to receive mailings from Digium.
On 11/14/07 8:47 AM, Philipp Kempgen [EMAIL PROTECTED] wrote:
Is the Whats New at Digium the Asterisk Company message I got from
[EMAIL PROTECTED] really from Digium?
IQ Labs announces the release of PBX Testing Framework.
This software is intended to test existing call-center PBX, and is
distributed under GPL license.
Currently it allows SIP testing, but implementing IAX (and even Zap)
shouldn't be a problem, as the framework is based on Asterisk, and can
do
I have been shaking down a dialplan for SIP fax to efax.
The basic senario is an ATA on the same subnet as the Asterisk 1.2 box
(avoid RTP packet lose and thus fax crash), calling a 'fax extension'
and envoking rxfax then email.
I leverage off of context: from-internal-additional-custom, so as
My calls provider has suspended my account, because he says that I am
send bad formating call strings.
According to the email he sent me a line return is beign inserted
after the number.
One string he sent me is the following:
BADCALL,101339,0115712550727
,reseller,cesar reategui
Hi,
I would suggest you to use Asterisk Application SIPGetHeader in your
Dialplan for incoming calls from plugandtel.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader
Something like
*
exten=_[a-z].,1,SIPGetHeader(Var_TO=To)
exten=_[a-z].,2,Dial(SIP/${Var_TO})
Please be
I had (I thought) a generic inbound route that was to handle all calls -
dumping them all into my queue after doing a bunch of time logic. Once
in the queue all the extensions are rung. The exception seems to be any
call that has blocked their callerid - those calls make it to the queue,
but
Doug Lytle wrote:
Nick Adams wrote:
The kicker: I can't seem to get past this 200 call point even though the
What does your console show at this time?
When testing, I've noted the 200 call limit was because I had too many
open files. I had to increase this by typing ulimit -n 4096
Every once in a while (like 2 out of 7 times), I get the following message:
[Nov 14 12:49:02] NOTICE[6855]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/5000-082508f0' not posted
I look in the cdr table in mySQL and indeed, the record is not posted for that
call.
This makes me want to create
Is there a way to enable the pip tones (beep) indicating that a call is being
recorded?
I know that ChanSpy does beep (unless q option is chosen) once, but not quite
the same.
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exten = _X.,1,Playback(beep)
exten = _X.,2,MixMonitor.
If you are starting the recording using some DTMF code sequence described in
features.conf make sure you use caller, callee or both value to play
sound to correct line end.
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing
On Nov 14, 2007 4:15 PM, Richard Cahilig [EMAIL PROTECTED] wrote:
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
cdr report. The message is when you open the cdr report is: - Call
Logs -
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
cdr report. The message is when you open the cdr report is: - Call
Logs - Back to Top
No data found !!!
1 / 1
Did I missed something in the
On Wednesday 14 November 2007 04:15:38 pm Richard Cahilig wrote:
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
cdr report. The message is when you open the cdr report is: - Call
Logs -
You need some experiance with the ANSI C programming language. Once
you have acquired that the rest is pretty straightforward.
On Nov 14, 2007 2:21 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
You mean modify the source? Could you give me an example, say I wrong
to remove advance option?
On
did you try
canreinvite=no
in your sip.conf file
It would also help to:
1) Post the relevant configuration files (phone AND Asterisk)
2) Post the EXACT message from column 1 to EOL
3) What version of Asterisk? Stock? From a certain distribution? Patches?
Or I could just say There is a problem
On Nov 14, 2007 6:31 PM, joakimsen wrote:
You need some experiance with the ANSI C programming language.
Once you have acquired that the rest is pretty straightforward.
http://www.amazon.com/C-Programming-Language-2nd-Ed/dp/0131103709/
--
___
Hi all,
I have an existing panasonic analog pbx in use and a asterisk server with
digium tdm400p(2 fxs and 2 fxo).
channel 1 - fxs - telephone
channel 2 - fxs - telephone
channel 3 - fxo - extension 15 at panasonic pbx
channel 4 - fxo - phone line from telco
We call in to fxo (channel 4)
On Wed, Nov 14, 2007 at 11:34:31AM +0800, Rilawich Ango wrote:
Hi all,
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
What about minivm?
--
Tzafrir Cohen
All looks fine to me - and hopefully nobody does anything nasty to your
server.
PaulH
On Thu, 2007-11-15 at 06:15 +0800, Richard Cahilig wrote:
Hi,
I installed asterisk-addons and asterisk-stats, Its working now except
of one problem. The problem is there is no call logs when you open the
Have a request from a customer to integrate Asterisk into an MS CRM
environment. From quick research it seems the way to do this is through
a CTI interface and it requires 3rd party middleware.
I have zero experience with MS CRM, so looking for some tips on if/how
this is possible, specifically
well.. if nothings working.. try putting in debug lines urself in the
code.. say
use system calls to write some debugging data into some temporary file
in ur perl code.
let us know..
Matt wrote:
[EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v
This is perl, v5.8.5 built for
On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote:
The Cisco Documentation states that you can modify standard and
nonstandard softkey templates. They may not be xml files. I just
assumed they were xml since that is what is used to configure the phone.
Just bumped into some info about
2007/11/14, Greg Oliver [EMAIL PROTECTED]:
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn't find much up there.
Thanks
Hi all
I want to create Ivrs using dialplan and aslo want to transfer call to
agent using Queue app in asterisk.
Is there any way to get IP ADDRESS of free agent which is found by asterisk
thnks ,
Bhrugu mehta
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