J. Oquendo wrote:
Stepping back into reality for a moment, emotions aside, I side with
the law if Vonage infringed on VZ's patent. Don't misconstrue what
I typed, re-read it clearly. If you created something, patented it,
and someone else used it without permission or compensation, if you
can
Due to our house layout I am unable to run some CAT5 cable from one
room to another. Therefore I purchased a Belkin wireless ethernet
bridge, but to my amazement it does not work :( Though, if I plug the
adapter into a PC ethernet port it works a treat. Connects to the AP
with a strong signal. I
On 22:23, Fri 06 Apr 07, Hans Witvliet wrote:
On Tue, 2007-04-03 at 05:30 -0700, Jason Kim wrote:
Is it exists?
If not, how could they have done this:
http://opensourcepbx.tmcnet.com/topics/applications/articles/5450-industry-forum-hails-successful-voip-over-asterisk-ipv6.htm
(But
Am Freitag, den 06.04.2007, 18:23 -0700 schrieb Am Turnip:
When I listen to voicemail from my Google Talk buddy, the envelope says,
from an unknown caller. But the voicemail correctly records the caller
ID of calls that arrive via Zapata into the same context that receives
Google Talk calls.
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still load the driver as modprobe qozap ports=12 as I've always
done. But now it
Hi
I have seen discussion about Asterisk integration with SugarCRM and
Salesforce.com CRM in mailing list archives.
I just want to add here that Star Outlook Dialer (Free Edition) has
built in integration (through StarJunction) with SugarCRM as well as
with Salesforce CRM. It is available
On Sat, Apr 07, 2007 at 12:17:03PM +0200, Henrik Woffinden wrote:
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2 ports instead of 4.
I still
Hi all,
Im trying dial a user according to the device s/he uses. i mean if the user
is using asterisk as a peer, then i have to pass the extension in the dial
application like this:
Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the
DNID.
and if the user is using
S. A. Kamran wrote:
I just want to add here that Star Outlook Dialer (Free Edition) has
built in integration (through StarJunction) with SugarCRM as well as
with Salesforce CRM. It is available for free download at
http://www.starutilities.com/staroldialer.htm
Can someone tell me how to
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
hi lee,
i have changed my config in iax, and now i can call make calls without
problem.
this is a description of my architecture:
i have two offices with SIP users having the extension _037XXX in the first
office and _022XXX in the second office.
i want to connect my two offices with IAX for
If your device is connecting to asterisk as a peer or a friend, the the sip
show peers user will show a user agent field. For example I have a
linksys phone in my home office that connects as a friend and so if I type
sip show peer 1000 - the phones username. I get the following entry
Useragent
Will an Outlook dialer run on Linux ?? Works fine on XP, might check your OS
:)
On 4/7/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
S. A. Kamran wrote:
I just want to add here that Star Outlook Dialer (Free Edition) has
built in integration (through StarJunction) with SugarCRM as well as
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:53 PM:
On Fri, Apr 06, 2007 at 12:37:47PM -0700, Lee Howard wrote:
It's usually built and left in the zaptel source directory where you
extracted and built zaptel. If it doesn't get built for you from
zttest.c then check the Makefile that
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10 based. IIRC, I never
found the file(s) needed to compile it.
Do you actually have a timing source?
head -c 0
Rizwan Hisham wrote:
Im trying dial a user according to the device s/he uses. i mean if the user
is using asterisk as a peer, then i have to pass the extension in the dial
application like this:
Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the
DNID.
and if
On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10 based. IIRC, I never
found the file(s) needed to compile
On 6 Apr 2007, at 21:21, Matthew Rubenstein wrote:
On Fri, 2007-04-06 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 6 Apr 2007 16:13:29 +0100
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage)
To: Jason Wolfe [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM:
On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe Acquisto wrote:
zttest does not exist on this system, Suse 10
Thanx a lot...it does it..i was in need of it very badly
On 4/7/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
Im trying dial a user according to the device s/he uses. i mean if the
user
is using asterisk as a peer, then i have to pass the extension in the
dial
On 7 Apr 2007, at 07:02, Jay Milk wrote:
If Verizon's patent claim is indeed so broad as to prevent Vonage's
PSTN interconnect, then Verizon would still have to show that the
patent is non-obvious and a truly new invention
Sadly not - if they have a patent granted, then the onus of
On Sat, 07 Apr 2007, Jay Milk wrote:
(my comments inline)
This sounds like you really don't know what these legal proceedings are
about. I googled this a little a week or two ago, when it appeared on
engadget of all places. It appears that VZ sued Vonage for infringement
of seven patents,
On Fri, 2007-03-23 at 03:11 +0530, Rajeev Natarajan wrote:
It happened again this evening and when I checked the host-id
in /var/log/asterisk/messages the time when it did not register, it
showed a host-id
Mar 22 18:14:48 VERBOSE[2586] logger.c: == G.729 Host-ID:
Eric ManxPower Wieling wrote:
I am experiencing the same thing. I assumed that I just didn't have a
fast enough CPU (2.4 Ghz Celeron Ghz, also tried on a 1.8 Ghz Pentium
4). I am using a T400P with an Adtran TA750 Channel Bank rather than
the Digium analog cards. I'm not doing any VoIP on
On Sat, Apr 07, 2007 at 08:42:59AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/7/2007 8:24 AM:
On Sat, Apr 07, 2007 at 07:47:12AM -0400, Joe Acquisto wrote:
Tzafrir Cohen [EMAIL PROTECTED] Wrote: 4/6/2007 9:52 PM:
On Fri, Apr 06, 2007 at 03:19:16PM -0400, Joe
I've dug down as far as I could on www.uspto.gov for
anything remotely close to what is going on with
Verizon and all searches end with only two
possibilities in regards to what is going on.
So unless the patent was issued to someone else and
Verizon bought it, these are the only two possible
J. Oquendo wrote:
So unless the patent was issued to someone else and
Verizon bought it, these are the only two possible
patents this case could be based on...
I am looking at Verizon Services Corp. v. Vonage Holdings Corp., Slip
Copy, 2007 WL 528749, E.D.Va.,2007, which is the result of the
They could be suing for patents completely unrelated to VoIP as a
technology. There are cases on the book where people like Katz have been
running around suing contact center operators because he has a patent on
authenticating yourself to a phone service using a pin number and using
that
Tzafrir Cohen wrote:
On Sat, Apr 07, 2007 at 12:17:03PM +0200, Henrik Woffinden wrote:
Tzafrir Cohen wrote:
On Fri, Apr 06, 2007 at 01:18:24AM +0200, Henrik Woffinden wrote:
Hello list,
After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently
detected as 2
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
. . .
So your timing source is basically working.
--
Tzafrir Cohen
And this means . . . any FAX-ing issues must be due to other problems?
joe a.
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khaled,
I successfull remaster a router CD, and lamppix CD both using knoppix
or debian as the base, I am pretty sure...
try
http://lamppix.tinowagner.com/
http://www.wifi.com.ar/english/cdrouter/
daveC
Khaled Chehab wrote:
Anyone have an idea to re master centos,in other
Hi All,
I am trying to upgrade an old Asterisk installation to 1.4.2 (it's
currently running CVS-08/02/04-15:15:26) but have hit a couple of
problems.
The first was easily fixed. I got storage size of sin isn't known
errors whilst compiling streamplayer.c, but after seeing
Jonathan Hunter wrote:
/usr/src/asterisk-1.4.2# make
[LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o
ast_expr2.o strcompat.o - aelparse
aelparse.o: In function `ael_yylex':
/usr/src/asterisk-1.4.2/include/asterisk/strings.h:167: undefined
reference to `__builtin_expect'
When my follow me or transferred calls come out to me they appear as if they
are coming from one of my lines rather than showing the caller id of the
initial caller. I believe there is a way to make it forward the initial
caller id information isn't there? Is it just that my voip provider is not
Anyone have an idea to re master centos,in other worlds I have an asterisk
on centos with all libraries and modules,how can I make it as an iso image
?
Have a look at Kickstart
hth
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Hi,
On 07/04/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Given the age of your distribution, I suspect your compiler is too old.
What version of GCC are you using?
I haven't compiled any Asterisk version on this machine since 2004, so
you could well be right on that front.
# gcc --version
Jonathan Hunter wrote:
Is __builtin_expect part of gcc, then, rather than an external
library? (i.e. would I need to upgrade gcc in this instance)
Yes, it is a GCC extension added in GCC 3.x, I believe.
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Zaptel has no direct code relationship with Asterisk. Your error is
because zaptel is trying to use a member no longer exists in newer
kernels. However you are using fedora, and fedora included that change
in older kernel. I found this in xpp/xbus-core.c
/*
* As part of the inode diet the
Here's links and descriptions for the 8 you listed. All Bell Atlantic, GTE,
or Verizon. This should make your research a bit easier.
6,137,869
Network session management
http://www.google.com/patents?vid=USPAT6137869id=yl4GEBAJdq=6137869
Patent number: 6137869
Filing date: Sep 16, 1997
Issue
Jonathan Hunter wrote:
The machine is quite old, so it is possible I need to upgrade/add
something - but what?
# uname -a
Linux myserver 2.4.25 #5 Wed Jan 26 18:20:35 GMT 2005 i686 unknown
# cat /etc/slackware-version
7.0.0
Slackware 7! Upgrade to something from this century! Seriously, 7.0
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre message dès mon retour. Pour toute urgence, contacter
Emmanuelle Parache Moga ou Cédric Buzay.
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To
joe,
when I have problems with audio and other connections seem to work, I
always look for a codec incompatibility... use 'sip set debug peer
extension' and look for the codec handshaking... make sure both
extensions have a compatible codec choice...
daveC
Using INVITE request as basis
Michael Boers wrote:
Thanks for the suggestion. I am using a tdm400p with 4 fxo channels. I
am in the US, so opermode should not be ok at default settings. I just
recently got fxotune working on my system. The version that comes with
zaptel 1.2.16 would simply hang. I am using the 1.4
Jay Milk wrote:
That last point could be quite a big one against VZ -- Vonage is gaining
customers not because they stole Verizon's doubtful IP, but because they
offer a better deal. In my area, Vonage is cheaper than a Verizon
dialtone alone -- and I'd still pay for each outgoing call if I
I need to authenticate users to make long distance calls. Basically,when
the user dials a long distance dialplan pattern, I want to prompt for his
pin and look it up against a table of pins:usernames in a file. If it
exists, I'll use the username in the cdr accountcode and permit the call.
Hi J French,
try with DISA ;p
http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA
--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
___
Stephen Bosch wrote:
Jay Milk wrote:
That last point could be quite a big one against VZ -- Vonage is gaining
customers not because they stole Verizon's doubtful IP, but because they
offer a better deal. In my area, Vonage is cheaper than a Verizon
dialtone alone -- and I'd still pay for
Has anyone tried pushing calls to a Vonage ATA attached to an FXO card in
Asterisk and had your account terminated by Vonage?
I'm curious as to whether they will stop your service if you push too many
calls through their ATA in a specific period of time.
Thanks in advance for the info, SG
Hi,
Someone knows and can explain what is wink, prewink, start and preflash
time?
Sds,
Gustavo
From: Gustavo Cordeiro [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: Re:
On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote:
snip
You are right. zapata.conf is not used in IAX connections. My reading has
led me to believe that manipulating gain on an IP PBX is neither necessary
nor practical in VoIP channels, so Asterisk does not devise such settings.
Thanks
There's no way for them to tell if you have asterisk on the fxo port BUT they
will terminate your account if you hook it up as the outbound for an office
pumping call after call through it. What did you expect?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
Hi,
I've had a hard time understanding what was going on in a new * setup.
The deployment has a * box running on dual xeon RH9 stock 2.4.20-8
and some different versions of asterisk (1.2.10/1.2.16) + libpri +
zaptel + wanpipe.
Short version: audio from iaxclient clients is fine from windows
but
If I were Vonage I'd have a delegation in HongKong now, moving all my
Telco interconnects
to somewhere where the US patent system is treated with the contempt it
is starting to earn.
ROTFL. The US patent system is treated with contempt in Hong Kong? You
have no idea how EXTREME
On Thu, Apr 05, 2007 at 02:06:53AM -0500, Jonathan Rivera wrote:
Hello all
I have this problem, i need a way to balance my trunks which are SIP
peers, when a SIP peer is busy then send the call for another peer and
so until i can send away the call, i think i can do it with queues.
Ok
Andy,
Who is your VOIP provider ? Off the top of my head you can try Teliax
(www.teliax.com), VoipJet (www.voipjetcom) and Nufone (http://nufone.net/). All
of these providers let you set your own CID.
Dovid
- Original Message -
From: Andy Gee
To: asterisk-users@lists.digium.com
Olivier,
You have two options.
1) Change the source code.
2) Pay a coder to give you the options.
Also this mat be the lack of sleep talking but from what I remember there was
talk about this before. Search the archives.
Dovid
- Original Message -
From: Olivier
To: Asterisk
snip
ROTFL. The US patent system is treated with contempt in Hong Kong? You
have no idea how EXTREME legislation in Hong Kong against IP 'theft' is in
Hong Kong.
/snip
I find this hard to believe since most hack attempts to my box's originate
from IP's in China.
If you want a specific CID to show up it seems that your only options are to
A) Write an AGI.
B) If you don't have many users that can dial international you can use a
series of GotoIf statements.
C) You can use the Asterisk DB.
Dovid
- Original Message -
From: J French
To:
Bob Smither wrote:
On Fri, 2007-04-06 at 21:42 -0700, Yuan LIU wrote:
snip
You are right. zapata.conf is not used in IAX connections. My reading has
led me to believe that manipulating gain on an IP PBX is neither necessary
nor practical in VoIP channels, so Asterisk does not devise such
You can set up a simple mysql table with PIN-users this makes it more
extensible and you can create a simple web interface to change to pins/add
users.
after you have set up the table just use a simple IVR construct to prompt
for the PIN, fetch it from the table and authenticate it - something
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