/wiki/BlindsideDevelopment
Richard
On 5/28/07, Dean Collins [EMAIL PROTECTED] wrote:
I think this is a great potential application for Asterisk – I couldn't
actually determine if/where you had a downloadable POC or if it was still
just in development conceptualization at the moment.
Either
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing
the instructions (build/deploy/etc.).
Code is located here.
http://www.blindsideproject.org:8080/cgi-bin/trac.cgi/browser
Hi Steve,
Yes, we are looking for that. Do you know of any projects that provides
those? I know one written in TCL/TK.
Thanks.
Richard
On 5/28/07, Steve Totaro [EMAIL PROTECTED] wrote:
Sounds cool. You could probably use some code from the various open
source jabber clients that allow
On 5/28/07, Roberto Fichera [EMAIL PROTECTED] wrote:
At 19.19 28/05/2007, you wrote:
On 5/28/07, Roberto Fichera mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
At 17.09 28/05/2007, Richard Alam wrote:
Yes, we have some downloadable code. We are in the process of completing
the instructions
not navigate away from the current page.
It requires an Asterisk Manager connection.
See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for more
details.
Kind Regards,
Richard Hamnett
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Cool, please send me the pattern so i can add it
Cheers
Rick
On 5/21/07, Alexandre VERNIOL [EMAIL PROTECTED] wrote:
Really Great!!! Works for me in France I have just change the pattern
and that's ok reallygood job!
Cheers,
Alex
Richard Hamnett a écrit :
Hi there,
Just to announce
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running an Asterisk based system, yet
Has abandoning SugarCRM, and opting to
Paul wrote:
Richard Lyman wrote:
Paul wrote:
Joe acquisto wrote:
Paul [EMAIL PROTECTED] Wrote: 4/30/2007 8:53 AM:
Joe acquisto wrote:
I have dual posted this to the user and biz lists.
Has anyone ever heard of someone running
callerid=Richard Klingler 995
mailbox=995
callwaiting=yes
transfer=yes
threewaycalling=yes
context=klingler
linelabel=phonelab
line = 995
any ideas left?
Using now cmterm-7970_7971-sccp.8-2-2SR1
cheers
rick
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: 32: Name=SEP00175A872053 Load= SCCP70.8-2-2SR1S
: Invalid SCCP message! : ID :82
Looks to me that chan_skinny doesn't understand many important messages.
Any previous 7970G SCCP firmware that might work?
cheers
rick
Richard Klingler schrieb:
Sorry bringing it up again
Meanwhile
Hello (o;
Did I miss somewhere the announcement of 1.4.3?
Also don't see anything in the announce mailing
list archive...but it is available for download...
So do I need to download to find out what has changed? (o;
cheers
rick
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try using this in zaptel.conf
span=3,0,0,d4,ami
*snipped
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this
application as that seems more for a provider - end user and Astbill
wants to control the workings/creating of users/peers or am I mistaken?
Thanks,
Richard
.
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be in
CVS-TRUNK as an option for chan_zap
Eric ManxPower Wieling wrote:
Richard Lyman wrote:
Eric ManxPower Wieling wrote:
I don't know where he got the bizarre
useincomingcalleridonzaptransfer option, but it does not exist as
you can see below:
*snipped
just a note, not sure if it is still in 1.4 tree, but it used to be
in CVS
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the
--- J. Oquendo [EMAIL PROTECTED] wrote:
richard Coco wrote:
Hi all,
i have asterisk 1.2.17 with sip tcp support and i
am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the
HG3540.
But when i try to place a call from Asterisk
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:* 9722/asterisk
972 is the tie access code from Hiapth to Asterisk.
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard
Coco wrote:
Hi all
sorry, it works with upd... I am now able to make and
to receive calls.
thx...
--- richard Coco [EMAIL PROTECTED] wrote:
strange i have:
udp0 0 0.0.0.0:5060
0.0.0.0:*
9722/asterisk
972 is the tie access code from Hiapth
Steve Edwards wrote:
*snipped
Psst -- don't tell the developers, but we could probably get something
similar to Asterisk with a box of tin cans, a spool of string and a
couple of carrier pigeons :)
don't forget the sneakers! G
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Hi,
I am configuring a conferencing server and need to
allow SIP clients guest access.
In iax.conf, I can allow guest access to the
[conference] context with this entry
=== iax.conf ==
[guest]
type=user
host=dynamic
context=conference
So anyone connecting without username/password will
Tried this...it worked...but is this the best way?
== sip.conf ==
[general]
context=conference
allowguest=yes
[guest]
type=friend
nat=yes
host=dynamic
canreinvite=no
context=conference
--- Richard OSS [EMAIL PROTECTED] wrote:
Hi,
I am configuring a conferencing server
Tomislav Parcina wrote:
Richard Lyman wrote:
*snipped
fyi: manager originate is
channel + context + exten + priority
OR
channel + application + data
not both.
So, you are saying that this should look like this?
Action: Originate
Channel: Local/[EMAIL PROTECTED]
Application: System
Data
Nathan Bell wrote:
I'm having trouble getting the manager interface to behave properly;
specifically the Originate event.
If I create an originate event as below, the calling phone will
auto-answer (as it's supposed to) but the receiving phone never rings.
It will timeout at 20 seconds.
Tomislav Parcina wrote:
Lee Jenkins wrote:
You have to login into the AMI server with proper credentials and
send commands.
*snipped
OK, maybe he doesn't show output, so I have tried this:
Action: Command
Command: ! rm /tmp/test.txt
Response: Follows
Privilege: Command
--END COMMAND--
But
Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote:
Hi Murphy,
Sorry! But I didn’t understand you :(
Can you give me an example?
When I talked about creating a new property in the events to return the
ActionID command, I just give an idea. My problem is to identify WHO has
raised
TP'n to follow flow.
'there were' means that *over time*
enough mods added to be able to track most of the 'call flow' by it.
referring to the callerid name manipulation method
Moacir O. de Souza Junior - Personalsoft Sistemas Ltda. wrote:
Richard,
When you said there were means
Alexander Lopez wrote:
It is a HUGE workaround but in concept it should work.
You will need to build completion confirmation into your script as you
will always get a success code from the manager.
Action: Originate
Application: System
Data: /path/to/script
Channel: Local/[EMAIL PROTECTED]
but doesn't actually ring any phones until a channel is available.
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED]
Number of Attachments: 0
This message (and any associated files) is intended only
Evnin'
As I didn't find any answer I'll try to rephrase the problem (o;
Any idea why the latest asterisk-addons-1.4 write wrong uniqueid
into mysql database?
Asterisk-1.4.2 creates call record files with the uniqueid
prepended:
1175107269-SIP-999-0876c000.wav
But into mysql
Evnin'
Now I tracked my problem down why ARI won't display most of
the recordings...
It write a recording for examples as:
1175031785-SIP-0615000995-0872a000.wav
But it writes to the field uniqieid into MySQL database as:
1175031779.16
WHen I overwrite the uniqueid field
does appeal more to me. The
closest one I have found so far is MOR from
http://www.kolmisoft.com/index.php?option=com_frontpageItemid=1 however
it doesn't seem to support mISDN (using the B410P BRI cards).
Does anyone have any experience with these or others?
Regards,
Richard.
.
Best
Afternoon
A little off-topic...but...
Does any1 know why recorded call with IAX2 in the filename
are not displayed within ARI?
LittleJohn's website isn't a helpful place for ARI (o;
cheers
rick
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Evnin'...
Anybody got an idea where those CLI messages come from?
[Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response:
Remote host can't match request NOTIFY to call
'[EMAIL PROTECTED]'. Giving up.
Interestingly all are caused by local IP used by asterisk-1.4.1
cheers
rick
Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;
I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP and local * which then
connects to outside * vis IAX...
cheers
rick
Hermann Wecke schrieb:
Richard
features available in proprierary system
as good as original ;-)
PJ
Richard Klingler wrote:
Hmm..interestingly no one answered if chan_skinny works with 7970G
on * 1.4.x (o;
I know that CIsco phones are bad with NAT and SIP...old story (o;
THat's why I use local Cisco phones with SIP
Bill Hackensack schrieb:
On 3/21/07, *Richard Klingler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
As chan_sccp is pretty much dead, doesn't compile on FBSD anyway
and isn't supported on * 1.4.x I tried going with chan_skinny...
chan_sccp is far from dead and it works
Evnin' (o;
As chan_sccp is pretty much dead, doesn't compile on FBSD anyway
and isn't supported on * 1.4.x I tried going with chan_skinny...
The Cisco 7970 registers and is being acknowledged by * but that's it...
I see no lines on the 7970 display configured and it is not reachable
or it
Hi all,
according to
http://sip.fontventa.com/content/view/15/44/ i have
compiled the mpeg4ip libries without problem. After
copying the app_mp4.c file into de Asterisk apps
directory and changing the Makefile like.
[...]
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@
wrote:
*snipped
If I can't be confident enough in an important source of information like
this then I can't be confident enough to provide an Asterisk solution to
businesses. That's the way I see it. Yea, it's a wiki but it's the best
source of info out there.
*snipped
sorry to see you
can use the
variable ${CALLERID(number)} .
- Original Message -
From: richard Coco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 13, 2007 10:53 PM
Subject: Re: [asterisk-users] voicemail scenario
Hi all,
i need help to implement a voicemail scenario. What i
am trying to do is the following.
user X dials a direct access for user Y voicemail and
is asked to enter a number (e.g 12345678) and then
leaves a message. Then asterisk sends a notification
with attachement. The problem is that i
Hi,
i finally managed to get it work using GlobalVar.
I still have a question. I have several context in my
voicemail.conf like
[default]
[customer_1]
[customer_2]
[customer_3]
How can i set a different emailsubject for each
context?
thx
--- richard Coco [EMAIL PROTECTED] wrote:
Hi all
)
Basically im using Zap/4 as a failover for a SIP trunk when thats not
available
the problem is at s,4 it just dials that number and never times out
any ideas
Cheers
Richard Trenchard
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Eric Bishop wrote:
show dialplan keeps showing contexts created by AEL. I tried
deleting /etc/asterisk/extensions.ael but kept getting these messages
in the Asterisk log:
Feb 14 21:39:53 WARNING[6074] pbx_ael.c: Unable to open
'/etc/asterisk/extensions.ael': No such file or directory
Feb 14
TP'n to follow flow
just like DNS, the 'root servers' would still see the high request hits,
prior to passing off to local caching app.
and *someone* must have this expense/headache to maintain them.
Natambu Obleton wrote:
Why not make it like DNS and have each provider have their lookups
Joe Greco wrote:
TP'n to follow flow
just like DNS, the 'root servers' would still see the high request hits,
prior to passing off to local caching app.
and *someone* must have this expense/headache to maintain them.
No, the root servers wouldn't. Please take a few moments to learn
Yuan LIU wrote:
On my wild learning curve, I encountered numerous occasions when a
channel remained in Congestion state after a Congestion() step
without going to the next step, which is Hangup(). I couldn't find a
definite pattern but it seems to happen when a channel is hung up by
the
Benny Amorsen wrote:
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL everytime you make a dns request, i agreed that it does not hit
RL the root servers, but every time you request a NON-cached one you
RL DO.
Nope. If you request foo.com and you have up to two days earlier
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL
TP'n to follow flow
or mod the /etc/asterisk/indications.conf
the /xxx is the duration (iirc)
example: busy is like 400/400,0/400
the /400 (each) is the duration
Eric ManxPower Wieling wrote:
Read the zapata.conf.sample file that comes with Asterisk
[EMAIL PROTECTED] ~]# grep toneduration
ManxPower Wieling wrote:
/etc/asterisk/indications.conf has nothing to do with the length of
DTMF tones sent out FXO ports.
Richard Lyman wrote:
TP'n to follow flow
or mod the /etc/asterisk/indications.conf
the /xxx is the duration (iirc)
example: busy is like 400/400,0/400
the /400 (each
Can someone comment why only Digium cards still under warranty are
eligible to use this EC at no cost, versus older cards?
Regards,
Richard
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extension and end up in each others
voicemail.
After I replace the m in the option string it works as per normal
again but with the moh playing.
Thanks,
Richard
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED
Yuan LIU wrote:
From: younss azzayani [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 17:58:08 +
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory
Pavel Jezek wrote:
I can confirm,
commands after Wait() are never executed in 'h' extension
and wait seconds argument in wait() is completely ignored
it's bug or feature? ;-)
h = {
NoOP(before ${EXTEN});
Wait(5);
NoOP(after ${EXTEN});
}
*snipped
in ael use WaitExten(5);
about executing diaplan when entering 'h'
extension, ie. after user hangs up phone...
and seems, something strange with processing wait() app in processiong
'h' extension in diaplan - timeout specified is ignored, and dialplan
stops processing
Richard Lyman wrote:
*snipped
in ael use
Rich Doughty wrote:
Richard Lyman wrote:
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set
*snipped
ast_set_variables(chan, vars);
insert pbx_builtin_var here --
ast_pbx_run(chan);
since DIALSTATUS and HANGUPCAUSE are both protected, you will probably
have to create another such as FAILEDCODE.
i hope
Rich Doughty wrote:
i am creating call files, and catching successfully the ones that don't
connect in a 'failed' extension. can anyone tell me how to find out the
reason for the failure (ie busy, no answer).
${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't
used) and
more, I would like to find out, as the local
Panasonic agents have not been much help.
Regards,
Richard
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Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Excellent little script. Thanks, Yehavi.
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED
to both the users and connect them.
Any ideas on how to achieve this will be appreciated.
Thanks,
Richard
.
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Hi all,
i have ser and asterisk on the same box with a public
ip address. When an UA behind NAT registred on SER try
to call the Voicemail or another UA registred on
Asterisk i have one way audio (caller cannot hear the
callee).
[UA/SER]--[router/nat]--[SER/Asterisk]
UA has private
How (and where) could you provision those phones ?
Do you have any support from Siemens or anyone ?
We have a HiPath4000 V1.0 interconnected with Asterisk
using oh323. I have flashed several OptiPoints (from
the HiPath) to SIP firmware. But again OptiPoints seem
to work well with Asterisk but
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Does anyone know if misdn and the B410P is working yet in kernel
2.6.18/19?
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL
Douglas Garstang wrote
*snipped
cat = 0x81507e0 mcao_QMain
tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds
*snipped
a quick run through of of app_queue.c (my copy) for anything directly
dealing with a reload
shows tmp in use for realtime
later a reference for convert to
lenz wrote:
Hello list,
one of our clients is going to be deploying a system with over 200
differently composed queues and 100 agents. We are going to do a full
test of the viability of this solution before deployment, but I was
wondering if anyone has experience of such a setup and if there
Hi,
http://www.communications.siemens.co.uk/enterprise/products/optiPoint_410s.htm
rich.
--- Olivier [EMAIL PROTECTED] wrote:
Hi,
Is anyone aware of a wired sip hardphone supporting
802.1x authentication ?
I've been told some Avaya and Alcatel ip phones
supported 802.1x.
As 802.1x
FreeRadius)
a howto about 802.1X Port-Based Authentication are
avalaible at
http://tldp.org/HOWTO/html_single/8021X-HOWTO/
2007/1/2, richard Coco [EMAIL PROTECTED]:
***
This message was sent to your KasMail disposable
email address:
Asterisk Users Mailing List - Non-Commercial
Discussion
*snipped
Second, when using a .call file (or the manager interface's Originate
action) the 'Dial' action is executed BEFORE entry into the dialplan, so
if it fails, nothing in your dialplan is executed and you get a somewhat
*snipped
not *exactly* true.
you need to add
;this extension
Benny Amorsen wrote:
RL == Richard Lyman [EMAIL PROTECTED] writes:
RL grr, i hate when i typo (and reply to my own posts) exten =
RL s/,2,Set(CALLERID(name)=OUTSIDE NAME|CALLERID(num)=xx)
Heh, if you want to chase typos, perhaps you should add an underscore
before
Douglas Garstang wrote:
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 20, 2006 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Match a Numer - then continue with,
dialplan
Douglas
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies. When someone
calls a number in their own
Richard Lyman wrote:
Douglas Garstang wrote:
-Original Message-
From: David Gomillion [mailto:[EMAIL PROTECTED]
*snipped
David, this is completely different from what I am trying to do.
Let's try this a different way. Let's say you have two companies.
When someone calls
to contacts from Outlook?
And if so how well does it work?
Thanks,
Richard
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED]
Number of Attachments: 0
This message (and any associated files) is intended
on a specific Silicon Labs
chip, which is used in the TDM400 FXO modules.
No X100P uses this chip, (and the chips they use do not have the feature
used), so fxotune does nothing.
Regards,
Richard
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Richard
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Hi all,
I recently installed asterisk 1.2.4 on a HP DL140 G2 server and co-located it.
My only problem with the box is that there
is a noticeable delay in the processing of agi scripts compared to any other
install of asterisk I have.
Has anyone got any ideas why this is happening and any
asterisk
randomly play them back with the moh? It would be easy because then we
could just update the files every month or whenever we need to.
Thanks,
Richard
Best Regards
Richard Soderblom
Network Configurations
Cell:
E-Mail: [EMAIL PROTECTED
Steve Murphy wrote:
*snipped
I've been fixing manager bugs here and there, and am willing to take on
any manager issues out there, for 1.4, and trunk, especially, so as to
have things nice and solid for 1.4 before it gets out of beta.
*snipped
Richard-- I'll lab up
1.4 and see if I can get
Hi all,
We're looking at replacing our ailing Avaya phone system, potentially
with *. I've been asked to draw up costs and so on. The advice on
hardware seems to be sketchy, generally, or I'm looking in the wrong place!
Douglas Garstang wrote:
The Asterisk Manager Interface is driving me nuts.
Whoever wrote it should be drawn and quartered.
Sometimes the data comes back separated by \r\n, and sometimes it's separated
by \n.
The whole thing is completely inconsistent, and trying to write any kind of API
for
James Texter wrote:
Doug,
Your issue isn't with the manager. It's with the CLI output you are
trying to hijack via manager :D If you run sip show peer 2944093 in the
CLI, you'll see a blank line, followed by a line that is * Name. It
appears what you really want is a manager Action to
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Richard Lyman [EMAIL PROTECTED] wrote:
just wait till you get a 'hiccup' that causes a line to get cut off,
drop a char, and continue on next line. G
(examples below)
I've made heavy use of the Manager interface for over 2 years
-Useragent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098
Reg-Contact : sip:[EMAIL PROTECTED]
ChanVariable:
parkstart,10
Doug.
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 3:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Eric Bishop wrote:
You can't hanup channels with a call file you can only create them no?
*snipped
actually you could hangup a call using a call file
example
Channel: Tech/Dev-occurance
Application: Hangup
Data: somecausecodevar or digit equiv
in results with or without the flag.
Does anyone have any idea why this isn't working?
--
Richard Cook
[EMAIL PROTECTED]
T: 705-223-2000 ext 2010
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Gregory Duchatelet wrote:
Hi all,
Another question for today, hope an answer for this one…
I have a program talking with asterisk via the AMI. I receive events,
and I would like to insert some events in the dialplan, which could be
catch by my program.
Any idea how to do this ?
Greg
Michael Collins wrote:
I’m interested in knowing if anyone else has worked around this issue:
I have an application that needs to check the status of the calls
going through Asterisk about every 5 seconds or so. I don’t want to do
“asterisk –rx ‘show channels verbose’” at the Linux command
Tzafrir Cohen wrote:
*snipped
Note that you better not use a terminal server settings. The SIP client
should run on the thin client's CPU, not on the server's CPU. The server
can help with the boot process (maybe a shared NFS root will prove
useful).
*snipped
that particular unit is also
I would have to second the Sangoma buy. Their tech support
is second to none and more then helpful.
I've never had any problems with their products that wasn't my own
fault.
/R
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
KennedySent: Tuesday, October 17, 2006
Richard G. Cavanna
Information Technology Manager
SyChip Inc.
P - 972.202.8840
F - 972.633.0327
You can buy a pre made breakout box or go directly to a patch panel.
I have used this one form VoIP supply with success
http://www.voipsupply.com/product_info.php?products_id=1164searchid=111
839
))
{
hmap.get(((HangupEvent)event).getChannel().substring(0,
8)).setIcon(new ImageIcon(personal_green.png));
}
}
}
}
thx in advance!
--- Tim Panton [EMAIL PROTECTED] wrote:
On 4 Oct 2006, at 16:33, richard Coco wrote:
Hi all
Jay R. Ashworth wrote:
*snipped
The ability to detect precise SIT tones on placed calls would be
*really* good.
actually it is damn near impossible.
in a perfect world, if all the switch providers where adhering to ITU
spec on SIT's,
then it would be possible. they sad part is (at least
Hi all,
first of all sorry for the question. I know there is
an asterisk-java mailinglist but i am not subscribed
to this list and i am sure there are asterisk-java
guru on this list who can help me.
I am trying to get the status of a peer using
SipShowPeerAction. Unfortunately the getStatus
(peerEntryEvent.getStatus()); } } And you have to register to receive your eventHandler managerConnection.addEventHandle(mgrHdlr);HTH. richard Coco [EMAIL PROTECTED] wrote: Hi all,first of all sorry for the question. I know there isan asterisk-java mailinglist but i am not subscribedto this list and i am sure
but it is outdated and does
not apply to 1.2
Any help would be great
Richard
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files in the directory ( such as Master.1 or .2 ) as one might expect with traditional logging.Does anyone know what could be going on?Richard
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