Good Luck,
I went though this with Yahoo in the early 2000s. Their basic argument is
that their mark is included in your mark and they want your domain. They
are domain bullies. I went ahead and purchased your app because it sounded
pretty cool. I wish the best for you.
On Tue, Aug 11,
On Wed, Aug 12, 2009 at 01:28:17AM -0400, Dean Collins wrote:
This isn't asterisk related but I figure several developers on this list
have built apps for Twitter (or other 3rd party API's).
Just found out a few hours ago I'm being sued by Twitter
Reminder: there are alternatives:
Hi
You could also do it with one extension but set the call limit for the
extension in the sip.conf to something like
call-limit=3
Which would allow 3 concurrent calls to the one extension
Ish
Jimmy Ezell wrote:
Thanks for the help, I really appreciate the feedback.
I tried ringing them
Thank you Steve for your help,
but I could find in youy conf where you have defined outgoing
trunk for each sip extension.
Please comment.
On Tue, 11 Aug 2009 13:27:27 +0530 wrote
Hey here are the sample configuration. Create a trunk in sip.conf file, add a
registry string.Registry
I am using the phones quite successfully, though I have not tried non-English
menus.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, August 12, 2009 12:33 AM
To: Asterisk Users Mailing List -
good evening,
can anybody give the maximum dahdi calls that can do an asterisk server with
4 te420 digium cards and the max iax trunk calls that trancode to g729 or
g723
and if we can use the maximum tdm calls that the server can provide for 6
te420 cards and the iax trunks in a server that
Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe can suggest another CDR GUI ?
regards.
Harry
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Dear all,
I want to setup the incoming calls, that don't use authentication in
sip.conf file.
My configurations as follows,
[2000]
type=peer
host=dynamic
insecure=port,invite; (both)
context=Testing
But when I call '2000', I noticed the following message in Asterisk
CDRTool operates on CDRs generated by RADIUS servers into the standard
'radacct' schema, along with some custom attributes added by OpenSER.
CDRTool is designed for use with OpenSER, not Asterisk.
harry R wrote:
Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe
I am also using them quite extensively, but with English menus. I know that
the Locale files from Cisco do not come with the firmware, but usually as an
update for CallManager. There are a ton of languages that work with the
latest firmware, but I have no idea how to actually get the files from
CDRTool operates on CDRs generated by RADIUS servers into the standard
'radacct' schema, along with some custom attributes added by OpenSER.
CDRTool is designed for use with OpenSER, not Asterisk.
src http://www.voip-info.org/wiki/view/Asterisk+GUI
CDRTool
harry R wrote:
CDRTool operates on CDRs generated by RADIUS servers into the standard
'radacct' schema, along with some custom attributes added by OpenSER.
CDRTool is designed for use with OpenSER, not Asterisk.
src http://www.voip-info.org/wiki/view/Asterisk+GUI
On 13/08/09 1:30 AM, harry R wrote:
Hello
Anyone who have already use/configure Asterisk with CDRTool ?
Or maybe can suggest another CDR GUI ?
In my article the other day on Asterisk applications:
http://www.venturevoip.com/news.php?rssid=2184
I linked to Areski's CDR Stats (which I've used
I have setup my asterisk box using freepbx. I can call extension and make
outbound calls. the outbound calls drop between 10-30sec. we are using
bandwidth.com and they have logged our call. below is your bad followed by what
they say is a good call. I can't figure out where the problem is on
I've encountered this issue a couple of times and we managed to resolve
it by updating the sip phone and the router it was connected to both to
use their latest firmware.
I know it's not a definitive answer but I've never truly got down to the
heart of the issue as with us it would affect just
Hellos,
I am having issues with my meetme conferencing. When I dial the conferencing
number, It hangs after a few seconds.I have read somewhere that I need to
enable ztdummy, which I have done but still no changes.
Here is my log
~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43
Have you tried to replicate the problem (call from a to b 3-5 consecutive
times to see if same result)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Wednesday, August 12, 2009 10:34 AM
To:
D you are a genius!
Thank you very much, this does exactly what I want. Worked like a
charm.
Just a little extra information for the archive.
I changed my PhoneMacAddress.cnf file .cnf to have the phone
configuration lines listed in D's post.
I also changed my extensions.conf file
we have three phones hooked up right now. we have tried on all the different
phones and have called several different external numbers. all with the same
result.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 12 Aug 2009 10:48:32 -0500
Subject: Re: [asterisk-users]
So you have executed this call scenario: 1-a, 2-a, 3-a, 1-b, 2-b, 3-b, 1-c,
2-c, 3-c and got failure on each of the 9 calls? And then replicated on the
good call (1-a,2-a.)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Dean Collins wrote:
I received this email 30 minutes ago stating that Twitter is suing me??
Basically they feel that my application - www.MyTwitterButler.com
http://www.mytwitterbutler.com/ does the following.
*1/ That anyone using the API to auto follow people are breaching the TOS??*
*2/
Hi all
I am just confused as to why Asterisk appends a strange number after the
Channel and Extension number. This moght be obvious to some people but I
have no idea why??
Eg: IAX2/100-9123
Where does the -9123 come from?? and can I ditch it? I have made a work
around to substring it out using
D. Dante Lorenso wrote:
part of this is making a statement to get publicity, if twitter really
didn't like what you were doing they'd simply cut off your app accessing
their servers. But obviously that is not what its about.
Dean Collins wrote:
I received this email 30 minutes ago
Nowhere does the letter say Twitter is suing you. It is a cease and
desist letter.
I suppose their threat about further action at the bottom can be
reasonably surmised to mean that they might sue you in the future, but
that is a far, far cry from Twitter is suing me!
--
Alex Balashov
yup just did all the same results
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 12 Aug 2009 11:14:43 -0500
Subject: Re: [asterisk-users] call drops after a few seconds
So you have executed this call scenario:
1-a, 2-a, 3-a, 1-b, 2-b, 3-b, 1-c,
So a good call works on all 3 lines and a bad call fails on all 3? Are
there numbers that alternate between good and bad?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 12, 2009 11:39 AM
On Tuesday 11 August 2009 22:22:45 Seth Mitchell wrote:
Hello,
I recently completed a fresh install of Asterisk
SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the
voicemail application module to load. Output from debug shows:
Note that the version number is severely
Yeh I'm starting to learn the difference - sorry first time I've ever
been ceased and desisted lol, still learning the vernacular.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
On Wednesday 12 August 2009 11:21:02 Dáibhéad Antoine O'Reilligh wrote:
I am just confused as to why Asterisk appends a strange number after the
Channel and Extension number. This moght be obvious to some people but I
have no idea why??
Eg: IAX2/100-9123
Where does the -9123 come from??
Don't count yourself as out of the woods yet... They will at best make
your product inoperable by denying your existing client base access, then
may still come back at you. I'd be prepared to launch a second and
subsequent product that does the same thing if needed. Welcome to
Obamerica!
On Wednesday 12 August 2009 12:19:51 Danny Nicholas wrote:
Don't count yourself as out of the woods yet... They will at best make
your product inoperable by denying your existing client base access, then
may still come back at you. I'd be prepared to launch a second and
subsequent product
Could have just as well have said Busherica or Pelosimerica, but anyway,
sounds like Dean is up a proverbial creek. Not only will Twitter get their
way, but anybody who bought this product will probably get a refund.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
well the good call is from bandwidth.com as example. we haven't had a good call
form your office. they all fail. so i tried calling the same external number
from each extension the a different external number from all three extension.
they all fail. the guy at bandwidth.com just sent us that
I fail to see how Obama has ANYTHING to do with this.
Danny, please DO elaborate so that I don't have to go on believing that you're
a fool.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
This really has nothing to do with Obama. As for the Cease and Desist
order, I will certainly side with Twitter on this one. You have chosen to
use a trademarked name as part of your domain and offer a service that
violates the TOS that YOU agreed to when you created your Twitter account.
This
It's still a free country, so to speak. You're not the first or last person
to think I'm a fool, but Mr. Obama has ABSOLUTELY NOTHING to do with this.
This is a good object lesson for 3PD's (in my foolish opinion).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
,1250014321.23186,
Thanks for your help,
Best, regards.
Cristian.
Argentina.
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virus 4329 (20090812) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
On Wednesday 12 August 2009 12:50:46 Dpto. de Sistemas wrote:
Hi, Why do CDR second field, src field have a dest
The real src field is 9500.
Is a bug??
Example;
Q-aereos,1147938811,9500,outbound,1147938811,DAHDI/31-1,SIP/95
00-0de0ea60,Dial,SIP/9500|60|t,2009-08-11 18:12:41,2009-08-11
-users
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firmas de virus 4330 (20090812) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
__ Información de ESET NOD32 Antivirus, versión de la base de firmas de
virus 4330
On Wednesday 12 August 2009 13:28:10 Dpto. de Sistemas wrote:
My Q-aereos.cvs file load outgoing calls, and my extensions are 95XX, is
impossible here a incomming call.
Wait your reply. thanks.
Well, then, you have an error in your configuration, because that CDR
described an incoming call,
Hi all,
I'd like to setup a really lean Asterisk installation that essentially has a
full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan)
on a Digium TE207P adapter that all it does is convert the ISDN channels to
SIP/IAX channels. Then I would add this Asterisk
http://www.merriam-webster.com/dictionary/twitter
have you actually violated anything ?? IP my ass... you simply used their
published API and wrote an application which might be better then theirs,
simply reply by asking them to state explicitly, what parts of the terms of
services you violated,
2009/8/12 Jonathan Thurman jthurma...@gmail.com
I am also using them quite extensively, but with English menus. I know
that the Locale files from Cisco do not come with the firmware, but usually
as an update for CallManager. There are a ton of languages that work with
the latest firmware,
Shashi Dookhee escribió:
Hi all,
I'd like to setup a really lean Asterisk installation that essentially has a
full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan)
on a Digium TE207P adapter that all it does is convert the ISDN channels to
SIP/IAX channels. Then I
NOD32 Antivirus, versión de la base de
firmas de virus 4330 (20090812) __
ESET NOD32 Antivirus ha comprobado este mensaje.
http://www.eset.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009
Shashi Dookhee wrote:
Hi all,
I'd like to setup a really lean Asterisk installation that essentially has a
full ISDN PRI (ATT, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan)
on a Digium TE207P adapter that all it does is convert the ISDN channels to
SIP/IAX channels. Then I
I think I'm missing the beginning of this thread, but I had this exact problem
with a Call Manager going to two SIP providers, one of which was BW.COM.. I
don't know if it will help, since presumably you're using asterisk, but with
the call manager, the problem was that there was no
I know I'm missing something here (been a long day)...
How can I specify more than one channel in a call file?
I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1...
Thanks
Dave
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons
Sent: Wednesday, August 12, 2009 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks Danny,
I do have a dial cmd with multiple arguments in my normal outgoing context. I
guess my question really is:
How do I tell the call file using Channel: XXX to use my outgoing context
instead of Zap/g1/xx or sip/trunk_x/xx directly?
-Dave
From:
Ok. Here's how you would do that:
Channel: SIP/170 (some local extension)
CallerID: SIP/104 (another local extension)
MaxRetries: 1
WaitTime: 60
retryTime: 5
Context: your_context
Extension: s
This should create an extension call using your context. The context can
then dial out as
Context: is what the call is dumped into after it is answered, at extension
Extension:. I don't think it's related to how the call is placed.
I can dial the local extension SIP/170 but I'm not sure where that gets me.
Basically I want to have the same failover that I have for all other outgoing
Your'e wanting control of the call from a call file. The way to do that is
to call using a context instead of a Technology/number.
When you call SIP/trunk_1, you are using the default context and therefore
don't have any fallthrough options unless you wrote them into your default
context. If
If you use a Local channel to dial it then it will fall under the same rules
Channel: Local/numbertod...@the-context-you-want
This gets a CDR produced, it does pay to check everything works the same
but it should be fine
Cheers Duncan
David Gibbons wrote:
Context: is what the call is dumped
On Wed, Aug 12, 2009 at 12:39 PM, Olivier oza-4...@myamail.com wrote:
2009/8/12 Jonathan Thurman jthurma...@gmail.com
I am also using them quite extensively, but with English menus. I know
that the Locale files from Cisco do not come with the firmware, but usually
as an update for
Duncan and Danny--
Thank you! I believe the Local/ is what I was missing with ex...@context.
-Dave
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan Turnbull
[dun...@e-simple.co.nz]
Sent:
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