FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
Digium launches Asterisk VCC (TM) - a new virtual communication platform
for enterprises, the public sector and the home.
===
Asterisk 1.8 will contain a stunning new
Just can't wait for the live calorie counter! :)
l.
2010/4/1 Olle E. Johansson o...@edvina.net
FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
Digium launches Asterisk VCC (TM) - a new virtual communication platform
for enterprises, the public sector and the home.
We were hoping voicemail would become Tweets and that Tweets from your
bathroom scale could be sent as audio using calls files. I guess that
will be the next minor version?
/r
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Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO ports).
HTH,
Ioan.
On Thu, Apr 1, 2010 at 12:29 AM, Kosa k...@piradio.org wrote:
I have two linksys spa2102 and a sap9000 but as far as I know I need
something else to connect the asterisk box to the
Press Release
For Immediate Release
OfficeSIP Communications
http://www.officesip.com/
i...@officesip.com
OfficeSIP Communications Makes Its VoIP SIP Products Open Source
OfficeSIP Communications makes its two enterprise VoIP SIP clients
officially open-source. OfficeSIP Softphone and
I'm not quite sure what do you mean with MSC.
Anyway, I assume your environment is like
[PSTN (Public Switched Telephone Network)]--[DTM
Switch]---SS7 (PRI line)[Asterisk
Box]VoIP (SIP/IAX etc...)- IP net
If you mean MSC Mobile
Redfone uses and improved, in house developed TDMoE driver, officially
supported by same Redfone.
Redfone´s support site maintains tdmoe driver updated and certified to
operate in every zaptel and dahdi versions.
Txs
Jorge Churio
Redfone Communications
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You know there are 1st of april jokes, and there are evil 1st of april
jokes.
... I actually felt a bit nauseous
- - Tommy
Den 01. april 2010 07:53, skrev Olle E. Johansson:
FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
Hi all,
My problem boils down to these errors:
... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time
This is triggered by lines in extentions.conf such as:
exten = _X.,1,Dial(ZAP/g1/${EXTEN},,W)
The
On Thu, 1 Apr 2010, Tommy Botten Jensen wrote:
You know there are 1st of april jokes, and there are evil 1st of april
jokes.
... I actually felt a bit nauseous
Thought it was funny myself! Not as funny as the other one just posted
about the VoIP clients written in C# and .net under
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on his phone.
Note that name of A gets displayed on the B's phone fine, but this is
not what I want.
This works with Cisco Call
Try using sendrpid=yes on sip.conf
Regards,
Juan
Ondrej Valousek wrote:
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on his phone.
Note that name of A gets displayed on
Thought it was funny myself! Not as funny as the other one just posted
about the VoIP clients written in C# and .net under Microsoft though. Now
that was funny!
It was not a joke. What is wrong with c#,.net?
Best regards,
Vitali Fomine
--
Ondrej Valousek wrote:
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on his phone.
Note that name of A gets displayed on the B's phone fine, but this is
not what I want.
I found a sap400, wich is FXO and seems to work fine with asterisk. It
offers the posibility to plug 4 analog phonelines.
Thanks again.
Kosa
- Un mundo mejor es posible -
Ioan Indreias escribió:
Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO
On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
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Hash: SHA512
You know there are 1st of april jokes, and there are evil 1st of april
jokes.
Huh? Have you actually bothered reading the site it points to?
The date there clearly states
Figured out my issue. My contacts are in mac-addr-directory.cfg when it
should be in mac-addr-directory.xml.
When did Polycom switched from CFG to XML?
From: hin lee hi...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
thank you very much. this solved my problem.
greets
felix
Am 31.03.2010 um 22:52 schrieb Mark Michelson:
Felix Tiefenthaler wrote:
Hi list,
can anyone tell me how to reset/delete all modifications (personal
greeting message, personal name, ...) I made in my voicemail?
I just want
Hi,
I'm seeing a lot of Exceptionally long voice queue length errors in
my logs, and then I seem to have a problem
where Asterisk will drop the registration for a significant number of
phones (they go UNREACHABLE), but then they
come back approximately a minute later.
Is this some sort of load
On Thursday 01 April 2010 10:36:23 Vitali Fomine wrote:
Thought it was funny myself! Not as funny as the other one just posted
about the VoIP clients written in C# and .net under Microsoft though. Now
that was funny!
It was not a joke. What is wrong with c#,.net?
You probably should have
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
You know there are 1st of april jokes, and there are evil 1st of
april
jokes.
Huh? Have you actually bothered
On Thu, Apr 01, 2010 at 07:05:07PM +0100, --[ UxBoD ]-- wrote:
- Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
You know there are 1st of april jokes, and there
Back to your original point, this is NOT the day to post a real news item;
we all know that's Friday after 4:30 EST.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, April 01, 2010
make menuconfig does not show cdr_odbc as a selectable compile option. I
have compiled and installed both unixODBC and freetds from source and have
verified both successfully connect to my sql server. Both were installed to
standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
On Thursday 01 April 2010 13:36:00 Nathan Pryor wrote:
make menuconfig does not show cdr_odbc as a selectable compile option. I
have compiled and installed both unixODBC and freetds from source and have
verified both successfully connect to my sql server. Both were installed to
standard
It seems that asterisk-addons and one or more of Digium's licensed
modules such as res_fax_digium have a conflict that doesn't seem to
be documented anywhere I can find.
In a nutshell, asterisk14-addons-core has a fake provide for
asterisk-gplonly :
#
# core subpackage
#
%package core
Yes, after installation of the drivers I've tried:
./configure
./configure --with-unixodbc=/usr
./configure --with-unixodbc=/usr/lib
None turn on cdr_odbc. I'm at a loss here. Is there anything in the
configure output I should be looking for?
On Thu, Apr 1, 2010 at 2:10 PM, Tilghman Lesher
- Nathan Pryor nathanpr...@gmail.com wrote:
make menuconfig does not show cdr_odbc as a selectable compile option. I have
compiled and installed both unixODBC and freetds from source and have verified
both successfully connect to my sql server. Both were installed to standard
locations
Chris Miller wrote:
A comment in the spec file would have been nice... Does anyone know
if this a real technical issue, or simply a licensing conflict
between GPL and Digium?
It is not a technical issue; it is an issue because some of the modules
in -addons have licenses that are pure GPLv2
I'm doing some automated calling by putting .call files in the Outgoing
folder of Asterisk. I'm concerned this might be a stupid question, but I'm
pretty sure I've done my research well and I'm unable to come up with an
answer on my own.
I want to know: what happens to the .call files after the
I was missing libtool-ltdl-devel. Thanks!
On Thu, Apr 1, 2010 at 2:48 PM, Jared Smith jsm...@digium.com wrote:
- Nathan Pryor nathanpr...@gmail.com wrote:
make menuconfig does not show cdr_odbc as a selectable compile option. I
have compiled and installed both unixODBC and freetds from
People,
Anybody knows what mean this message in my CLI:
[Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary is
no more. He has ceased to be! He's expired and gone to meet his maker! He's
a stiff! Bereft of life, he rests in peace. His metabolic processes are now
Hint Apr 1
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Apr 1, 2010 at 6:05 PM, Marcus
Youd think that this is/was some kind of April fool message, but it is a
real 1.6 warning
http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html
Since 1.6 has more multi-thread capabilities, the good folks at
Digium/Asterisk made this warning program to keep runaway threads
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Hi, Alyed.
On Sun, 28 Mar 2010, Alyed wrote:
I didn't know that there was Digium's GUI. It is FLOSS? I was looking
for in the site of Digium in the download section, but the unique
thing that I saw that it speaks of a GUI is AsteriskNow, that in
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Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
I think if you are installing dahdi complete from source you do
make all and make install and make config
Something that I forgot to ask previously is if the update of Asterisk
or DAHDI is
Hi All,
I have a question about how a particular situation would work between two
PBX systems:
If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same
rack, same network), and then pass a call from the Mitel to Asterisk to
perform some functions (lookups, maybe routing), and
On 4/1/2010 1:52 PM, Kevin P. Fleming wrote:
Chris Miller wrote:
A comment in the spec file would have been nice... Does anyone know
if this a real technical issue, or simply a licensing conflict
between GPL and Digium?
It is not a technical issue; it is an issue because some of the modules
Depends on the configuration you make. For example, if you want to route the
call giving the Mitel a new desrination or prefix, you can use Transfer
dialplan app. Transfer before answering the call will be redirected with SIP
302.
If the call is to be anwered on *, then canreinvite set to yes
This begs the question of when the actual violation occurs. In other
words, is this really a usage issue, or does the violation occur
at install time even though the non-GPL component is not usable?
It's hard to see how the violation could occur unless and until the
resulting program were
On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
Depends on the configuration you make. For example, if you want to route the
call giving the Mitel a new desrination or prefix, you can use Transfer
dialplan app. Transfer before answering the call will be redirected with SIP
302.
If the call
I finally got it at Calories Consumed. Geesh. Good one! :)
j
On Thu, 1 Apr 2010, Olle E. Johansson wrote:
FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
Digium launches Asterisk VCC (TM) - a new virtual communication platform
for enterprises, the public sector and the
If * answers the call, it will be on the loop but with canreinvite or
directrtp the media can be out of * and redirected to the final end point even
if signaling goes through *.
For the trunk, you can have multiple simultaneous calls. I do not know about
Mitel's licensing but with only one
Danny,
I haven't been able to test further, but I've seen the same issue since
I upgraded to 1.6.2.6. The astcanary does not appear to stay up more
than a few seconds when asterisk is initially started. On this same
system using 1.6.2.2 (the previous version I was running prior to the
Hello,
i´ve got this when i asterisk has died / killed and was restarted but i
dont have seen that it will collapse then.
i also got this after restarting asterisk from the CLI with restart now.
so dont worry ;)
best regards
steve smith
Danny Nicholas schrieb:
You’d think that this is/was
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