are or software.
Brad Allen
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gt;>>>
>>>>> dtmfmode=info
>>>>>
>>>>> or
>>>>>
>>>>> dtmfmode=rfc2833
>>>>>
>>>>> Doug
>>>>>
>>>>>
>>>>> --
>>>>> Be
On Tue, May 21, 2013 at 11:26 AM, Matthew J. Roth 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
> >
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On Thu, Jun 14, 2012 at 4:05 PM, Kevin P. Fleming 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 &q
button display?
Thanks for all comments!
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en the '='
> and closing ')' less leading and trailing spaces is part of the
> name. For SIP this may be detrimental to the message format
> unless the quotes get escaped.
>
> Richard
>
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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 wrote:
> Just a hunch
> same => n,Set(CALLERID(name)="AB\&CD COMPANY"
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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 Mailin
mins', 'marked', or 'locked'.
> conf
> Conf refers to the name of the conference being referenced.
>
> Guess the developers of confbridge didn’t want to duplicate the meetme_count
> function?
>
> From: asterisk-users-boun...@lists.digium.com
> [m
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gards,
Baldvin
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royles
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hurs:
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ry and
get 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
ing about this and possibly replying with your
thoughts.
Sincerely,
Baldvin
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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
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may be something and I'll
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
&
using the
right terms for this or it is rarely documented?
Any details much appreciated.
Thank you!
Baldvin
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orporation, One Microsoft Way, Redmond, WA 98052
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orking reverse
in-addr.arpa lookups, 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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oblem, however they are
> not the brightest bunch :-)
>
>
>> regards
>> klaus
>>
>> PS: use the asterisk-video mailing lists
>
> Just have :-)
>
>
>
>>
>> Asterisk Users schrieb:
>>> Hi all,
>>>
>>> I am trying to
,tTkK)
exten => dialcell,n,Hangup()
exten => t,1,Goto(s,2)
sip.conf:
[general]
context=sip_in
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
videosupport=yes
disable=all
allow=ulaw
allow=alaw
allow=h263+
;allow=h263
;allow=h263p
allow=speex
allow=gsm
#include "/etc/pbx-ta
you at
> least have support. and 3 clone cards and hours of troubleshooting
> later you will wish you hadn't been all cheap.
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?
tnx,
Baldvin
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I'll continue with my replace-with-silence-file method for now. Thanks for
the input.
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that they are not applicable (like the periodic announcement),
but just to not spend time and resources on playing a file that we would
rather not hear.
Thank you for your clear reply though, you make an excellent point regarding
the existing configuration options.
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of view because then we get file-not-present
warnings.
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s-
> [EMAIL PROTECTED] 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/supporte
NT box <-> B410P port in TE mode <-> Asterisk <-> B410P port in NT
mode <-> PBX
Thank you for your time and effort!
Respectfully,
Baldvin
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aster
nuals (although that was specifically geared
towards their cards, I must say)?
Thank you for your time and effort!
Respectfully,
Baldvin
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very much appreciated.
tnx,
Baldvin
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9
To: [EMAIL 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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asterisk-users mai
scripting, external programs of some sort (runnin ubuntu 7.04 or 6.06) or
whatever works. I'd really appreciate your input here.
Sincerely, Baldvin
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aster
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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ne is ready to
listen. Would really appreciate your input on this.
tnx,
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 th
read this and perhaps respond.
Sincerely,
Baldvin
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= 0x0
DISCONNECT_IND ID=002 #0x0197 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3490
DISCONNECT_RESP 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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gt; seems to work well.
> Unfortunatly I do not have 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-
>
> ___
> --Bandwidth a
t; I can send you 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
> >
the
> > phone.
> > Has anyone else noticed these problems, or does anyone have a copy of
> > 1.1.0.5.
> >
> > -Drew-
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews
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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Anyone have or know where I can go to get the astcc
voice prompts in spanish ? Thanks.
Dovid
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Dovid
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:
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 Discussion<asterisk-users@lists.digium.com>Message-ID: <[EMAIL PROTECTED]>Content-T
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.di
Asterisk work when less
than two weeks ago you were publicly bashing on the list.
Steve
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randisSent: Monday, 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
8
Merry 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
22, 2005 8:18 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
ginal Message-
From: [EMAIL 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
a number and nothing happend. Just my 2 cents.
Regards,Dovid
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OTECTED] On Behalf Of Dakota
Sent: Tuesday, December 06, 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 g
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 i
aging Program.
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handle_request: Unknown SIP command 'PUBLISH' from '192.168.100.100'
By the way I’m using * 1.0.9 CVS-HEAD September
15. 2005
Best regards
BennyBad
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TED]
[mailto:[EMAIL 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,
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ear).
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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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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well with asterisk ? Thanks a lot.
Dovid
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management would prefer a php
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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A
to
something that is of NO use.
Any suggestions or recommendations is appreciated.
Best reg.
BennyB
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To
our provider's call
allowance. 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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insure they are refunded and happy with
the resolution.
Thank you,
David Deutsch, President
Tris Telecommunications, LLC
(800) 547-4057 x1001
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wondering if anyone else has
recent experience getting CON-SNT-CP7960 in the U.S.? What
retailer did you use?
Weird.
~Adam
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To
a page, which is how much the phone call costs to send a
one page fax).
Thanks for making it through my long post...I'd by happy to get any
answers about any part of what I mentioned above!
~Adam
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ou'd be better off price wise getting a
single T1 card 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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h
ate
Asterisk coding skills) for helping to implement 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 base
teps) to upgrade the older 7960's.
> >
> >
> Rich, I am telling what I saw on my 7940: it was in a constant loop
> trying to upgrade again and again.
>
> You may say that Cisco phones are easy upgradable and all one need is
> just follow instructions, but that
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ut 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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ia: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport
From: ""Nik Martin IAX" ;tag=as40e79563
To:
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Date: Wed, 03 Nov 2004 15:21:52 GMT
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Dear All
Is there anyone out there who is using a VoiceXML system with Asterisk?
Thanks
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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
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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*. 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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e the From-line to be the same as my incoming alias for
this list. <[EMAIL PROTECTED]>
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