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On Tue, May 21, 2013 at 11:26 AM, Matthew J. Roth mr...@imminc.com wrote:
asterisk users wrote:
I'm having a strange problem recently with a Yealink SIP-T28P phone
connected
to Asterisk 11.4.0 via openvpn. It was working fine for months, and now
when I
dial anything from the phone
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On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 06/14/2012 04:57 PM, asterisk users wrote:
We couldn't see anything about this on the Digium site, but maybe
someone here can comment?
Do the new Digium phones provide good teleworker functionality?
Yes, I
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Same result:
Executing [s@incoming:3] NoOp(SIP/choicetel-0092,
CALLERID(all): \ 519111) in new stack
On Fri, Nov 18, 2011 at 2:05 PM, Danny Nicholas da...@debsinc.com wrote:
Just a hunch
same = n,Set(CALLERID(name)=AB\CD COMPANY)
-Original Message-
From: asterisk-users-boun
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the meetme_count
function?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 11:10 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge 1.6.20 user count
Hi
, but I don't know if it returns a variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk users
Sent: Wednesday, November 09, 2011 12:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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this going.
tnx!
Baldvin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 6. júlí 2009 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What is the best way to share extension
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I think I got it. ${DIALEDPEERNUMBER} contains the leg that connected
(just what I need).
FYI I used DumpChan() to get all the available variables and found it.
Thanks!
Enlai
On Wed, 24 Jun 2009 14:19:46 -0700, asterisk-users@lists.digium.com
said:
Thanks Danny.
I tried accessing ${CHANNEL
terms for this or it is rarely documented?
Any details much appreciated.
Thank you!
Baldvin
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look closer into that as well as continuing to look for other ways to do
this.
tnx!
Baldvin
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: 26. maí 2009 19:42
To: novacks...@gmail.com
ation, One Microsoft Way, Redmond, WA 98052
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, and as you might have guessed, all mail would
originate from the local host.
Suggestions?
Thanks!
-Karl
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Dear asterisk-users@lists.digium.com, Best Price Only Today.
http://byz.domemax.com?itg
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-video mailing lists
Just have :-)
Asterisk Users schrieb:
Hi all,
I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.
I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried
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and hours of troubleshooting
later you will wish you hadn't been all cheap.
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installation that
requires this. So statically doing this is ok in my case.
I'll continue with my replace-with-silence-file method for now. Thanks for
the input.
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(although that was specifically geared
towards their cards, I must say)?
Thank you for your time and effort!
Respectfully,
Baldvin
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in NT
mode - PBX
Thank you for your time and effort!
Respectfully,
Baldvin
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] On Behalf Of Kevin P. Fleming
Sent: 17. nóvember 2007 23:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple B410P's in one machine
[EMAIL PROTECTED] wrote:
1) Is it possible/supported to install two or more B410P Digium cards
in one
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or 6.06) or
whatever works. I'd really appreciate your input here.
Sincerely, Baldvin
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PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to get TCP access to CDR Master.csv
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is ready to
listen. Would really appreciate your input on this.
tnx,
Baldvin
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that voice is in fact being passed through
Asterisk during the call (some counters etc.)?
Thank you for your time and effort to respond.
Baldvin
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TienSen Chong
Sent: 17. maí 2007 10:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Queue MOH
Is there any way if i want the caller to hear dial tone rather than the
MOH?
Perhaps you
this and perhaps respond.
Sincerely,
Baldvin
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ID=002 #0x0197 LEN=0012
Controller/PLCI/NCCI= 0x101
ISDN1#02: CAPI INFO 0x3490: Normal call clearing
== ISDN1#02: Interface cleanup PLCI=0x101
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a copy of 1.1.0.5 so cannot downgrade the
phone.
Has anyone else noticed these problems, or does anyone have a copy of
1.1.0.5.
-Drew-
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cannot downgrade the
phone.
Has anyone else noticed these problems, or does anyone have a copy of
1.1.0.5.
-Drew-
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this firmware, if you mail me off-list.
Bye
DV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GXP
Anyone have or know where I can go to get the astcc
voice prompts in spanish ? Thanks.
Dovid
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I would if the tech that sets it up knows exactly
what he or she is doing.
Regards,
Dovid
: "Dakota" [EMAIL PROTECTED]Subject: Re:
[Asterisk-Users] Asterisk vs 3COMTo: "Asterisk Users Mailing List -
Non-Commercial Discussion"asterisk-users@lists.digium.comMessage-I
:
15Date: Mon, 09 Jan 2006 15:28:28 -0500From: Ken D'Ambrosio
[EMAIL PROTECTED]Subject: [Asterisk-Users] "Decent" sub-$100 SIP phone.To:
Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO
rds,
Dovid
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Asterisk work when less
than two weeks ago you were publicly bashing on the list.
Steve
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, January 02, 2006 8:05 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Client SIP fo Windows Mobile
Hi all,
anyone known if is there any SIP
client to install on an I-Mate SP5m with Windows Mobile
?
Thanks
Giordano
Christmas everyone!
Regards
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: Saturday, December 24, 2005 7:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Busy signal for incoming
:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] What hardware fits my needs?
Hello:
I want setup an asterisk based VoiceMail Server(IVR).
I have around 20K distinct users(DIDs) dialing to my system through
telephones/mobiles. The users can dial to their mailboxes and listen
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Wegrzyn - asterisk
Sent: Tuesday, December 20, 2005 8:31 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] dtmf problem
hi everyone,
I do have 2 lines with broadvoice.
From 2 days on one line my dtmf tones are not passed
a number and nothing happend. Just my 2 cents.
Regards,Dovid
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, 2005 5:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk vs Nortel, Northstar and Mitel
How does Asterisk compare to Nortel, NorthStar and Mitel PBX systems?
For a medium size company not growing past 75 extensions, would you
recommend Asterisk
You'll find the GSM codec renders smaller filesizes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, November 16, 2005 12:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Recording voice messages in mp3
.
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: Unknown SIP command 'PUBLISH' from '192.168.100.100'
By the way Im using * 1.0.9 CVS-HEAD September
15. 2005
Best regards
BennyBad
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PROTECTED] On Behalf Of Paul Hewlett
Sent: 19. september 2005 18:49
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] hints and the sNOM 360
Hi
I am trying to get a SNOM 360 to monitor other extensions i.e. when
someone
makes a call to/from another extension, one of the LED's
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).
For reference, all traffic is being sent through using G.729, but I
don't think that this makes any difference.
Many thanks
John
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http
business DSL. Thanks.
David
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Hi,
Already posted once but I need more feedback. What
kind of servers is everyone using for asterisk and what problems have you ran in
to ? Thanks.Dovid
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with asterisk ? Thanks a lot.
Dovid
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based solution.
What have other admins done to retrieve detailed call information
about the queue system? Anyone develop their own that they don't mind
sharing?
--johann
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to
something that is of NO use.
Any suggestions or recommendations is appreciated.
Best reg.
BennyB
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. Check with provider for call costs.
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, I have no documentation at all on how to program the 6 soft keys.
It's not explained in the User Manual. Anyone know how? Thanks.
-Ron
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they are refunded and happy with
the resolution.
Thank you,
David Deutsch, President
Tris Telecommunications, LLC
(800) 547-4057 x1001
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else has
recent experience getting CON-SNT-CP7960 in the U.S.? What
retailer did you use?
Weird.
~Adam
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...I'd by happy to get any
answers about any part of what I mentioned above!
~Adam
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and a channel bank.
Otherwise, the Voicetronix board may or may not even do the 1k/s.
--
Mike Mattice - Systems Programmer and Administrator
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an after hours w/
optional override feature, similar to what this poster mentioned:
http://lists.digium.com/pipermail/asterisk-users/2004-November/071460.html
I think it'd be great if we had the option to conditionally include
contexts based *either* on time [the existing capability] *or* based
need is
just follow instructions, but that won't help the man who was pulling
his hair off.
Andrei
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of
order? Shouldn't zaptel be loaded first?
Maybe my original install is a little hacked. Where do you load all your
modules and asterisk from on startup of your server?
I have a T100P and a TDM400P installed.
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Dear All
My business provides Asterisk consultancy in the UK. I am traveling to
San Diego / Tijuana from the 4th to the 13th and wondered if there were
any fellow Asterisk users who would like to meet for a coffee / drink?
Please email me direct ([EMAIL PROTECTED]).
Regards
John
Dear All
Is there anyone out there who is using a VoiceXML system with Asterisk?
Thanks
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).
The Maximiser only speaks H323 (not SIP) and can act as an H323
Gatekeeper, so in theory we should just be able to log on to the system
(he said, wishing it was that simple!).
Many thanks
John
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of *. Unfortunately, when compiling there are lots
of errors, many of them relating to the non-existent /usr/src/linux
directory.
Is there something special which I should be doing?
Thanks
JB
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I saw somewhere that the last kernel to work properly with the zaptel
drivers when using data over it was 2.4.20. Has this been since fixed to
work with newer kernels?
-Mike
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,Goto(iaxtel-guest,s,1)
exten = 17006691152,s,Goto(isdn-free-dial-out,,s,1)
unfortunately it doesn't seem seem to work easily, maybe because IAXTEL
doesn't send me the called ID ?
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