Ex Vito [EMAIL PROTECTED] writes:
I don't have access to an asterisk system right now
(nor any other sort of information source) but I seem
to recall that from 1.4 onwards the config option for
recording queue calls is named differently...
Is it mixmonitor ? Check you 1.4
Please check:
http://bugs.digium.com/view.php?id=12112
which we had to fix ourselves. There are still problems using:
1. patterns in extensions
2. queue members
3. sip.conf, iax.conf, voicemail,etc should all work fine. Note the
schema include with the distribution is invalid for the supplied
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I am using the rxfax and txfax application with Asterisk 1.4.18. When
ever I try sending or receiving a fax, my Asterisk dies. I tried to
enable debug to see what happens, but I have no clue why it happens.
Please help me out.
--
Regards,
Nasir.
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On Thu, 2008-03-06 at 08:21 +0100, randulo wrote:
On Thu, Mar 6, 2008 at 5:32 AM, Carole Migden [EMAIL PROTECTED] wrote:
Generally what you know is best
This is close to the best advice I've seen on this list in the last 5
years! The rest is a question of religion ;)
Should have read:
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On Thu, Mar 6, 2008 at 9:42 AM, Hans Witvliet [EMAIL PROTECTED] wrote:
Should have read: The Future Of Telephony (asterisk bible)
It says the same already in the first edition of the book
Actually, it's fairly common wisdom outside of mailing lists and IRC :)
Hi,
I have a setup where asterisk (1.4.18) is connected to a SIP proxy. Now
the SIP proxy challenges REGISTER and INVITE request from Asterisk. Asterisk
is able to handle REGISTER request challenge but for INVITE request, it
seems to handle authentication for only one user that exists under
Hi,
I have just checked again and the Solaris build of the codec appears to
be v33 and not v34 as advertised.
Thanks
Bruce
Bruce McAlister wrote:
Hi,
The Solaris build still appears to be at v32. Am I being a little hasty :)
Thanks
Bruce
The Asterisk Development Team wrote:
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SUMMARY:Declined: VoIP
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Thank you all for answers. As I understand s - i and others is device specific.
I will not need them in my SIP configuration.
2008/3/5, Andres Jimenez [EMAIL PROTECTED]:
On Wed, Mar 5, 2008 at 1:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This is not needed. If the extension is not found,
On Thu, Mar 6, 2008 at 11:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
If you want to reply to this message regarding the schedule,
please reply to the author. Your messages look very badly in the
archives. And there is really no need to have 500 replies to this
message on-list.
Sincere
Hello list,
I'm having some problem integrating the SELECT TIMEDIFF() and SELECT
DATEDIFF() in my code. The syntax I'm using works without any problems if I
run them directly from the MySQL Client, but from the Asterisk Dialplan it
just wont work. Is there a limitation in the MySQL() application
Hi
On Thu, Mar 06, 2008 at 09:29:01AM +0100, randulo wrote:
Every week we try to get guests with ideas, products and services you
haven't had time to check out to come and talk about what they're
doing.
Tomorrow, Pika Technologies will be with us.
Friday, March 7that 12:00 PM (Eastern
Lee, John (Sydney) wrote:
Has anyone encountered such problems before?
On the IP501 and IP301, yes. The handset cord dies. I've had to
replace 3 so far.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
context testsql {
s = {
MYSQL(Connect connid ${DBHOST} ${DBUSER} ${DBPASS} ${DB});
MYSQL(Query resultid ${connid} SELECT
TIMEDIFF(callend,callstart) FROM tblCall WHERE id=7);
MYSQL(fetch fetchid ${resultid} temp);
MYSQL(Disconnect ${connid});
}
}
/CODE
The error
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Hello,
[Mar 6 07:07:51] WARNING[9994]: chan_iax2.c:3904 iax2_trunk_queue:
Maximum trunk data space exceeded to ***.***.***.***:52213
I am seeing a ton of these errors on an IAX2 trunk to a second server
with only 1 call on the trunk. I have found some information regarding
the MTU size
Is there a way to set up a user/peer in iax.conf where it matches
incoming calls based entirely on IP?
I have a provider that sets the username (as well as the extension) to
the phone number that has been dialled, I'd prefer calls from that
provider to all be identified as the same trunk.
TIA
On Thu, Mar 06, 2008 at 08:43:43AM -0500, OCG Technical Support wrote:
I (like many others probably have) added the sender of the invite to my spam
filter. That avoids the many replies - and also blocks future email from
someone stupid enough to spam multiple entire list with an invite!
The
Hi all,
In the changelog of bristuff, as of version 0.4.0test4(test5) the
beronet cards should be supported.
Can anyone confirm if the beronet 2,4 and 8 ports version are
supported by qozap now?
Regards,
stoffell
___
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I (like many others probably have) added the sender of the invite to my spam
filter. That avoids the many replies - and also blocks future email from
someone stupid enough to spam multiple entire list with an invite!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Evan Ruff wrote:
Since when is the users list a transport for calendar scheduling?
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth
Hi -
Thank you all for answers. As I understand s - i and others is device
specific.
I will not need them in my SIP configuration.
The s extension is not zap-specific. You can use it for any type of
device. It's just the generic extension that a call will go to when
no other matching
Greetings list,
I've been asked to provide a system for 200 extensions, most of which will be
existing analogue POTS handsets, not IP handsets. I've not really had any
experience with large channel banks in the past (since most of our deployments
are strictly IP-only to the desk), so I'm at a
Doug Lytle wrote:
Evan Ruff wrote:
Since when is the users list a transport for calendar scheduling?
Since when are humans infallible? Randy made a mistake. He apologized
for it. Let's move on...
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
Hey screwed up and has already apologized for it ok.
Randy tried something and didn't realize all the replies were going to
be 'resent' to the list.
At least he's out there trying something different. And as he's
tirelessly promoting asterisk every Friday afternoon putting his own
time and
Chris Bagnall wrote:
Greetings list,
I've been asked to provide a system for 200 extensions, most of which will be
existing analogue POTS handsets, not IP handsets. I've not really had any
experience with large channel banks in the past (since most of our
deployments are strictly IP-only
On Thu, Mar 06, 2008 at 03:21:47PM -, Chris Bagnall wrote:
I've been asked to provide a system for 200 extensions, most of which
will be existing analogue POTS handsets, not IP handsets. I've not
really had any experience with large channel banks in the past (since
most of our deployments
Actually, UNIX [tm] Describes meeting a standard, and not development
history.
http://en.wikipedia.org/wiki/Unix#Branding
Absolutely! Which is why I referred to Linux as Unix-like and not UNIX.
Linux is NOT licensed to use UNIX(r) per The Open Group's specs.
BSD and Mac OS X are licensed
On Thu, Mar 06, 2008 at 09:39:05AM -0600, Bill Andersen wrote:
Actually, UNIX [tm] Describes meeting a standard, and not development
history.
http://en.wikipedia.org/wiki/Unix#Branding
Absolutely! Which is why I referred to Linux as Unix-like and not UNIX.
Linux is NOT licensed to
Dean Collins wrote:
Hey screwed up and has already apologized for it ok.
Please note the time the message was sent. 8:58AM
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote:
www.citel.com
I used them a few years back in a pilot install with legacy Nortel
phones and it worked well. I gather they have grown tremendously from
there. I'm in North America, don't know how well they support UK stuff.
Citel
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What is the format of the UNIQUEID variable?
It seems to be something like:
40651204817492.56
Does it always have the pattern
long_number.short_number?
Be a better friend, newshound, and
know-it-all
On Mar 5, 2008, at 5:46 PM, [EMAIL PROTECTED]
wrote:
If you are running a call centre (large or small) using Asterisk,
I'd be
interested to know how you log your agents in out:
E.g.
- Do you use AgentLogin (to force calls onto the agents, perhaps)?
- Do you still use
On Thu, Mar 6, 2008 at 10:49 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Mar 06, 2008 at 10:38:36AM -0500, Drew Gibson wrote:
www.citel.com
I used them a few years back in a pilot install with legacy Nortel
phones and it worked well. I gather they have grown tremendously from
Hi, I need to interact with my Asterisk and need a good Java class library.
What do you think is the best?
Thanks
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To UNSUBSCRIBE or update options
On Thursday 06 March 2008 10:07:26 Vieri wrote:
What is the format of the UNIQUEID variable?
It seems to be something like:
40651204817492.56
Does it always have the pattern
long_number.short_number?
UniqueID is composed of the epoch when a call starts, plus a monotonically
incrementing
I have Asterisk 1.4 tied via SIP to a Cisco Callmanager 6.1 system. Calls
between the systems (ie. extension to extension) work perfectly.
However when I attempt to make an outside call from an Asterisk extension
through Call Manager to the outside world, it connects but only for a few
seconds,
In article [EMAIL PROTECTED],
Vieri [EMAIL PROTECTED] wrote:
What is the format of the UNIQUEID variable?
It seems to be something like:
40651204817492.56
Does it always have the pattern
long_number.short_number?
If the system has been running a long time with many calls, it could
be
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I believe I have all the necessary packages installed.
Having done some research, one link suggests using strace and in that
case I
don't get the error:
strace -f -o /tmp/trace -e trace=process ./configure
...
configure: *** Zaptel build successfully configured ***
That's from
--- Craig Guy [EMAIL PROTECTED] wrote:
I believe that IAXVAR in Asterisk 1.6 will do what
you want. I have a
backport of this for Asterisk 1.2.14 or so floating
around somewhere but it
hasn't been maintained or used for months, may not
be compatible with the
1.6 implementation and I
Hi,
I would like to seek an opinion or list of providers in USA or particularly
in California. We would need someone who can offer maximum ports and lowest
rates.
Thanks very much,
Vivek
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I have an Asterisk server with voicemail(), in the sip.conf I have:
[general]
allowguest=yes
language=pt_BR
I have the sound files for pt_BR in /var/lib/asterisk/sounds/pt_BR, and the
others dirs (dgits, phonetic and so on). The problem I have is: when a guest
tries to place a call and is
Vivek,
What do you need, DID or Termination?
BTW We are in California. Send me you Contact info and we can discuss more
about your needs.
-Jai
On Thu, Mar 6, 2008 at 10:25 AM, Vivek Shrivastava [EMAIL PROTECTED]
wrote:
Hi,
I would like to seek an opinion or list of providers in USA or
On Thu, Mar 06, 2008 at 06:50:40AM +1300, CSB wrote:
When attempting to build zaptel I get the following error:
configure:2184: error: C compiler cannot create executables
Where do you actually get the error from? From the 'make' command? If
so: go chase errors in menuselect/configure
--
I am using version 2.2.0.
__Yehavi:
Date: Thu, 6 Mar 2008 15:01:26 +1100
From: Lee, John (Sydney) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT How to Change Polycom Web Admin User:Pass
As far as I recall
Asterisk-java, http://asterisk-java.org is a very good one, it has a pretty
good documentation.
On Thu, Mar 6, 2008 at 1:01 PM, equis software [EMAIL PROTECTED]
wrote:
Hi, I need to interact with my Asterisk and need a good Java class
library.
What do you think is the best?
Thanks
When attempting to build zaptel I get the following error:
configure:2184: error: C compiler cannot create executables
Where do you actually get the error from? From the 'make' command? If
so: go chase errors in menuselect/configure
./configure
Cameron
Hi all,
I am wanting to use an option from the ./configure script with zaptel to
compile zaptel for a different kernel than the running kernel.
How do I do that exactly.
Example:
Current kernel is 2.6.18-8.1.4.el5
and I want to compile zaptel for 2.6.18-53.1.4.el5
I am using centos 5.1 or RHEL
What do we want NET NEUTRALITY
When do we want it? NOW AND FOREVER
This video should be compulsory viewing for everyone in public office
not just here in the USA but globally so this public resource cant be
stolen from you !!
http://deancollinsblog.blogspot.com/2008/03/net-neutrality.html
Jerry Geis wrote:
I am wanting to use an option from the ./configure script with zaptel to
compile zaptel for a different kernel than the running kernel.
How do I do that exactly.
Example:
Current kernel is 2.6.18-8.1.4.el5
and I want to compile zaptel for 2.6.18-53.1.4.el5
I am using
i've been using *1.2 w/ realtime static zapata in mysql table
fine. but after i upgraded to 1.4. it seems like the zapata
table doesn't load correctly. i have to go in the console
and use the zap restart to get the zap channels register.
is this sounds like a bug or something i'm missing when
Hi Asterisk-user, Steve;
I´m using *libmfcr2-0.0.3.tar.gz, libsupertone-0.0.2.tar.gz,
libunicall-0.0.3.tar.gz,spandsp-0.0.3pre22.tgz* with Fedora core 6
,Asterisk 1.2.14, libpri 1.2.4 , zaptel 1.2.20; So everything is working
perfectly with MFCR2, but sometimes i have problems with
On Thu, Mar 06, 2008 at 03:19:21PM -0500, Jerry Geis wrote:
Hi all,
I am wanting to use an option from the ./configure script with zaptel to
compile zaptel for a different kernel than the running kernel.
How do I do that exactly.
Example:
Current kernel is 2.6.18-8.1.4.el5
and I want
Hi,
Are those messages from [EMAIL PROTECTED] messages from
Digium sent via some kind of spamming service?
I did not subscribe to anything at en25.com so why
would I have to unsubscribe?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's
--- Tony Mountifield [EMAIL PROTECTED] wrote:
In Asterisk 1.4 or later, an optional system name
can be defined in
asterisk.conf, and if defined, the unique ID
becomes:
system_name-timestamp.seq_num
Thanks!
So for the sake of backward compatibility, if I dont'
define sysname in 1.4 then
On Thu, Mar 06, 2008 at 11:02:50AM +1100, Paul Hales wrote:
And we found (recently) that if you send the right http packet to a snom
phone you can make the screen say Agent 155 rather than the extension
number. :)
Or, y'know, INSERT COIN.
Cool New Website For everyone to see!
I think they are using a specially programmed version of Asterisk to do
this.
www.dialaway4free.com
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To
Dear Michael;
This problem happens even if I am in Zapata level (did
not use any SIP trunk), it happens when I am calling
to the asterisk box and need to enter the extension,
then it reads the digit duplicated.
Any advise?
Regards
Bilal
--
I believe you need to set in the
I can't believe I fixed the problem, but here's what I did:
1. Checked the Use Media Termination Point in the profile for the SIP
trunk in Call Manager.
2. Split the SIP config for Call Manager into separate inbound and outbound
settings like so:
3. Added the insecure=very to the callmanout
Apart from the BS crap about patent pending - looks like a great service
and I'm sure they'll get a ton of traffic.
Good use of technology.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From:
I have been told to use Rhino Channel Bank but I am yet to set it up and
I appreciate if someone can show me some doco of using Rhino on an E1/T1
with TE410.
Thanks.
I've been asked to provide a system for 200 extensions, most of which
will
be existing analogue POTS handsets, not IP handsets.
Hi guys,
I have made a upgrade to my addpac ap200c, however it does not upload
complete, now I can load addpac. Is there anyway that can I upload the old
firwmare? Any help is appreciated.
System Boot Loader, Version 2.2.5/DUAL(852)
Copyright (c) by AddPac Technology Co., Ltd. Since 1999.
yeah,
/Searching US Patents Text Collection.../
*Results of Search in US Patents Text Collection db for:
dialaway4free*: 0 patents.
No patents have matched your query
Original post also sounds a bit spammy to me... *shrug*
Brooks R. Bridges
Telecommunications Manager
Ifbyphone, Inc.
Phone:
Hi guys,
I have made a upgrade to my addpac ap200c, however it does not upload
complete, now I can not load addpac. Is there anyway that I can upload the
old firwmare? Any help is appreciated.
System Boot Loader, Version 2.2.5/DUAL(852)
Copyright (c) by AddPac Technology Co., Ltd. Since 1999.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What kind of information are you looking for? configuration or? If you
look in our manuals our cards and the Digium cards configure the same
in zaptel and zapata.
Lee, John (Sydney) wrote:
I have been told to use Rhino Channel Bank but I am yet to
Ex Vito wrote:
On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote:
Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very
likely, 380's as well). I just learned this the hard way.
--J
...can you expand on that please ? I'm on my way to
I think they are using a specially programmed version of Asterisk to do this.
Don't you mean:
I am using a specially programmed version of Asterisk to do this.
?
domain: dialaway4free.com
created: 16-Jan-2008
last-changed:
I've been asked to provide a system for 200 extensions, most of which will
be existing analogue POTS handsets, not IP handsets. I've not really had
any experience with large channel banks in the past (since most of our
deployments are strictly IP-only to the desk), so I'm at a loss as to
On Thu, 6 Mar 2008 17:54:04 -0500, Dean Collins wrote:
Apart from the BS crap about patent pending - looks like a great service
and I'm sure they'll get a ton of traffic.
Good use of technology.
Y'think? I have no patience for such adverts. It even bugs me to have
to listen to the Talkshoe
We had a similar issue where the connector was not pushed in hard
enough.
I know that sounds like a joke, but it isn't!
PaulH
On Thu, 2008-03-06 at 18:27 +1100, Lee, John (Sydney) wrote:
I have been testing with Polycom IP600 phones for a month or so.
I found out that there are frequent
On Thu, Mar 06, 2008 at 11:50:43PM +, Gordon Henderson wrote:
I've been asked to provide a system for 200 extensions, most of which will
be existing analogue POTS handsets, not IP handsets. I've not really had
any experience with large channel banks in the past (since most of our
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system.
I know the subject line was anti-Dell, but just to put in a data point:
We have 10 Dell PE2950's running with one or two TE220B's per system, and they
have
What kind of problems are you talking about and what you want to modify?
On Thu, Mar 6, 2008 at 2:42 PM, Jessica Gonzalez Arriagada
[EMAIL PROTECTED] wrote:
Hi Asterisk-user, Steve;
I´m using libmfcr2-0.0.3.tar.gz,
We have found that 860's with Te120p's seem to work well too.
PaulH
On Thu, 2008-03-06 at 19:29 -0500, Ron Joffe wrote:
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system.
I know the subject line was
I'm head of RD for a dot com company and we are looking to create a
prototype using asterisk. Basically we people who visit our site and search
for goods listed by other people. Once something is found, a phone number
is listed in the results and person A calls person B to see if the item is
I think tilghman hacked out something like this in less time than it took
me to search through 20 pages of googlegook trying unsuccessfully to find
it :)
A caller calls host A. They select a service provided by host B which is
invoked using dial(iax2/[EMAIL PROTECTED]/${EXTEN}|2|g).
The
We had a similar issue where the connector was not pushed in hard
enough.
I know that sounds like a joke, but it isn't!
PaulH
Thanks Paul - it also happened to my phone!
Thanks so much.
___
-- Bandwidth and Colocation Provided by
blackwater dev wrote:
I'm head of RD for a dot com company and we are looking to create a
prototype using asterisk. Basically we people who visit our site and
search for goods listed by other people. Once something is found, a
phone number is listed in the results and person A calls
I really see this is useless since we alreadu got pricegrabbers
buy.com and froogle they all list the itme in stock on the site there
is really no need for a $30k a year operator to read it for the
person.
just my $0.02
On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote:
I'm head of RD for a dot
I think he's talking about an automated system. It's definitely
possible with asterisk, whether or not it's a good idea.
I really see this is useless since we alreadu got pricegrabbers
buy.com and froogle they all list the itme in stock on the site there
is really no need for a $30k a year
It's certainly possible, and I would be interested in helping you get it
going.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of blackwater dev
Sent:
It was one of those moments in life where I felt a lot less smart than I
usually do...
PaulH
On Fri, 2008-03-07 at 15:28 +1100, Lee, John (Sydney) wrote:
We had a similar issue where the connector was not pushed in hard
enough.
I know that sounds like a joke, but it isn't!
PaulH
Hi,
I have an astrisk pbx installed on my system and i have registered
two Aastra hardphones and one SJPhone(softphone) with that. Then i tested
the following scenario
A(Aastra) calledB(Aastra)
B answered the call
I pressed conference button on the A ( A put B on
On Fri, Mar 7, 2008 at 12:58 AM, Michael Graves [EMAIL PROTECTED] wrote:
Y'think? I have no patience for such adverts. It even bugs me to have
to listen to the Talkshoe self-promo stuff when I miss a VOIP Users
Conference and download the MP3 recording.
Download and scrub. Repeat. :)
On Thu, Mar 6, 2008 at 7:25 PM, Vivek Shrivastava
[EMAIL PROTECTED] wrote:
I would like to seek an opinion or list of providers in USA or particularly
in California. We would need someone who can offer maximum ports and lowest
rates.
The usual suspects IMO (random order): Teliax, Nufone,
I was successful to control the max users (10) if I hardcode the
conference room number (in this case 101) as follows:
exten = 8600,1,Playback(conf-thereare)
exten = 8600,2,MeetMeCount(101)
exten = 8600,3,Playback(conf-peopleinconf)
exten = 8600,4,MeetMeCount(101,CONFCOUNT)
exten =
Gonzalo,
Please let us know what you mean by 'stops working' - it should spit
out errors or wrong queries to ldap.
Also please keep this list in your replies. I have no problems
answering personal emails but both of us might get more feedback if we
post our progress on the list! :)
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