ABBAS SHAKEEL wrote:
why don't you post your question
On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi motamed...@gmail.com
mailto:motamed...@gmail.com wrote:
On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com
mailto:csi...@gmail.com wrote:
Sunday, January 10,
Rick Green wrote:
On Thu, 7 Jan 2010, David Gibbons wrote:
Yes, gmail DOES default to top posting, because bottom posting is silly
(in general, but especially for a client that hides quoted text (like
gmail)). Top posting is modern. And better. And doesn't make me scroll
through 10
Steve Totaro wrote:
read your posting and it will tell you haw to remove yourself.
On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com
mailto:ric.d...@gmail.com wrote:
Can I be taken off the mailing list please.
Thanks.
rick
Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
Lately I have been using Thunderbird with an RFC2369 header plugin.
--FP
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
jonas kellens wrote:
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid
extension '11', but no rule 'i' in context ...[snip]...
When testing IVR and pressing 1 from my Grandstream SIP-phone, the
above message is printed on the Asterisk CLI.
How come Asterisk receives my 1
Alex Samad wrote:
On Tue, Dec 15, 2009 at 08:59:34PM +0100, Benny Amorsen wrote:
Gavin Spurgeon gspurg...@dageek.co.uk writes:
iSip (£2.39)
http://itunes.apple.com/gb/app/isip-push-service-formerly-sipphone/id298202722?mt=8
I have been very impressed by the audio quality
Joseph wrote:
On 12/08/09 11:11, Jared Smith wrote:
On Mon, 2009-12-07 at 19:39 -0700, Joseph wrote:
After pressing *1 console is not showing anything indicating that the
call is being recorded:
I find that I often have to adjust the featuredigittimeout setting in
Dr. Michael J. Chudobiak wrote:
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
on the PCI bus.
Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue
Francesco Peeters wrote:
Francesco Peeters wrote:
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from 82.101.62.99:5060 ---
SIP/2.0 180 Ringing
Allow:
Francesco Peeters wrote:
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
--
--- SIP read from 82.101.62.99:5060 ---
SIP/2.0 180
John F. Ervin wrote:
What do you do if you find things sharing interrupts (IRQ 11) in my
case with my X100P card. I believe there is some sort of internal
audio card in my cheap slow PC.
Check the BIOS whether you can:
Change the IRQ assignments
Disable the extra hardware using the same IRQ
Tzafrir Cohen wrote:
On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much processing power)
Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for
need to configure mISDN correctly as well! And
AFAIK you will need to use PTMP, as that is what the router would expect...
--
Francesco Peeters
Ubuntu all the way!
1 laptop, 1 server, 1 desktop at home
and several servers in different locations
would expect...
--
Francesco Peeters
Thanks for clarifying I've double-checked that it is running ptmp but still no
link lights. Anyone got other suggestions?
Regards
Lee
Are you using an ISDN cross cable? I don't know these cards, but most
cards are wired as a DTE type
the
convertor to get proper CID
If the Dutch mailing list starts I will join ;-)
Erik de Wild
Tripple-o
Me three! ;-)
--
Francesco Peeters
Ubuntu all the way!
1 laptop, 1 server, 1 desktop at home
and several servers in different locations
plan, and the only thing you achieve
on a more expensive plan is to pay less per unit, but flat-rate is
NON-EXISTANT...
It is one of the few things I actually envy my US colleagues for! (Of
course, we do have more PTO! G)
--
Francesco Peeters
Laptop: IBM T43 with Ubuntu Gutsy Gibbon, Workstation
for a potential client and it was hard to tell the TTS
bits from the human bits. If I took the time to learn Cepstral's markup
language I probably could have fooled myself :)
Thanks in advance,
Are there any tools like these for Dutch language Asterisk installs?...
--
Francesco Peeters
no sigs
All these repeated list replies with Autoreply: Autoreply: Autoreply:
Autoreply:... subjects are irritating at best and debilitating at worst!
This makes the list waste bandwidth and my inbox (and the archives too)
unreadable!
Thx!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1
On Fri, May 11, 2007 08:21, Gordon Henderson wrote:
On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote:
If you think your ISP is reliable enough then go for it!
I've had less ADSL issues last year than ISDN issues! ;-)
(And that while ADSL is running over that very ISDN line
On Fri, May 11, 2007 10:31, Chris Bagnall wrote:
There is a small (and growing!) number of small businesses (and not so
small ones either!) who are moving towards using their broadband
(typically ADSL in the UK) connection for Telephony - and even
installing
a 2nd ADSL line just for VoIP.
I forgot to pay this month's phone bill, and never noticed until family
(the in-laws, who are too cheap to try the cell phone if landline fails,
because it is 'more expensive') told me they were unable to reach us...
As it turns out, the phone company disconnected us, but because Asterisk
routes
On Thu, May 10, 2007 23:44, Gordon Henderson wrote:
On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote:
It gives me pause though... Maybe it's time to get rid of my fixed
line...
;-)
No ;-) needed - I have friends on cable internet with no separate copper
phone line now.
I'd
On Fri, May 11, 2007 07:34, Armin Schindler wrote:
On Thu, 10 May 2007, Crazy Boy wrote:
Hi Friends,
Can anybody tell me other softPBX softwares like Asterisk?
- OpenPBX
- Freeswitch
Or try Googling for something like 'open source pbx'... Sheesh! :-o
--
F Peeters
PIII 450 - 1 GB - *
On Sat, March 24, 2007 19:10, Bruce Reeves wrote:
You might get a faster response on freepbx/amp mailing list.
On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote:
SNIP
Just an update:
Still have NOT been approved for either the mailing list *or* the forum!
I am pretty
On Thu, March 29, 2007 19:36, Carlos Jerónimo wrote:
Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx
foruns this week, and my login is inactive yet. In the mail i receive
this msg:
Welcome to FreePBX Forums Forums
Please keep this email for your records.
Hi all,
I am probably missing something ultimately obvious, but I have a problem
configuring freepbx...
Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu
installation guide on freepbx.org.
System pxe-boots from a server with NFS root on same
Using * 1.2 current (from source, not
On Sat, March 24, 2007 11:54, Mauro Zanin wrote:
Hi everybody
I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded
software.
I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN
CARD
in a normal Italian EUROISDN installation. The * works fine except for
On Wed, October 18, 2006 19:03, Paul Gaffney wrote:
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution and my SIP phones are good but not
dependable.
Neil
Neil,
www.asteriskguru.com http://www.asteriskguru.com/ lists a few of
them.
On Wed, October 18, 2006 21:07, Guillermo Salas M. wrote:
On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote:
On Wed, October 18, 2006 19:03, Paul Gaffney wrote:
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote:
Hi all
I didn't change anything that's my point
It has be running and working just fine then at 4:32 pm yesterday I
could not make or recieve VoIP calls via our VoIP Provider
They say the Invite packet was being rejected and thus there
On Mon, September 4, 2006 16:55, Cory Andrews said:
Please be aware that from a future support standpoint, you may be a bit
limited with Zultys. Their future seems very uncertain they have recently
just about ceased operations and let the majority of their employees go.
Cory J Andrews
On Mon, July 31, 2006 21:44, Tom said:
At 02:21 PM 7/31/2006, you wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
Tom Vile wrote:
Did you look on the site?
http://www.4psa.com/products/voipnow/demo.php
Does above means that the
...
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
___
--Bandwidth and Colocation
On Wed, June 28, 2006 10:14, [EMAIL PROTECTED] said:
Well, look at it this way: if you get the working, you can buy one of
those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia
soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
interface
On Tue, June 27, 2006 0:26, shadowym said:
They have been talking about this for awhile. If you look at the real
time
and embedded operating system world they have not really done so well over
the many years they have been trying. Just throwing money at the problem
has
never worked for them
On Mon, June 26, 2006 20:06, Brian Capouch said:
Tzafrir Cohen wrote:
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote:
Marco, bom dia.
Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo
externo?
É freeware?
Podemos seguir com o projeto Asterisk-PT?
English,
On Mon, June 26, 2006 21:39, Brian Capouch said:
Francesco Peeters (Asterisk) wrote:
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig
Engels praten!
;-)
Pues my punto fue que un poquito de correo en otro idioma no hace daño,
y si ayuda mucho y molesta poco, ¿por
On Mon, June 12, 2006 4:37, David Choo said:
I will be out of the office starting 12/06/2006 and will not return until
17/06/2006.
Dear Sir / Mdm,
I'm currently travelling.
During this period of time, I have minimal access to internet and email.
As
such, please be aware that I might
On Wed, June 7, 2006 14:09, Louis-David Mitterrand said:
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote:
Well, these are encouraging words :)
You're basically telling me that I should tell my client to buy other
phones. I agree that you cannot compare these phones with Cisco or
On Mon, May 29, 2006 16:20, Remko Muis said:
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
On Wed, May 24, 2006 10:16, El Flynn said:
[EMAIL PROTECTED] wrote:
We have some laptop soundcards that are really bad and I would be glad
if you could share your experiences when changing to a USB headset
instead of using the built in soundcard in your computer.
Well, IMO if the
Labtec no longer sells USB headsets...
Good luck!
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate
for receiving
faxes = disabled *should* disable fax detection by causing it to use a
different branch of the AMP macro's...
HTH!
--
Francesco Peeters
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
Hi All,
Any VoipBuster SIP users on this list that'd be willing to test
VoipBuster outbound VoIP to PSTN?
All numbers I tried from my (*) server are supposedly being connected,
but no phone rings!
Also their new WebStart function doesn't cause my phone to ring either...
TIA!
--
Francesco
)
HTH!
--
Francesco Peeters
PIII-450 - 512 MB RAM - 2x HFC-PCI - BRIstuff Florz patch
AMD Duron 1GHz - 512 MB RAM, 2x HFC-PCI - vISDN
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
, and a session is established, but the
remote host never receives the command.
I *think* it has to do with the command shell environment in which the
system command is opened...
Any suggestions on how to set this up would be appreciated...
--
Francesco Peeters
PIII-450, 512 MB, 2x HFC-PCI
Francesco Peeters schreef:
Hi,
I would like to execute a command on a different system using ssh.
When I execute the command from the CLI on the asterisk machine, it
works fine (I set up RSA keys on both sides)
When I execute the same command from System() inside the dialplan, the
log
the actual call gets connected, ie immediately
after the channel proceeds from setup to actual ringing...
Is there any way of making this happen, and preferrably with as little
change to the [EMAIL PROTECTED] macro's as possible?
Any ideas would be appreciated...
--
Francesco Peeters
No sigs
On Fri, March 24, 2006 12:01, Asterisk said:
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no
On Tue, March 21, 2006 16:51, Mimmus said:
Hi,
at my Asterisk box, I have a few of IAX2 phones (configured with
alaw/ulaw/gsm codecs, in this order) and a PRI E1 line.
In iax.conf I hav:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
During some incoming call, I read at console:
On Wed, March 22, 2006 0:06, Steve Kennedy said:
On Tue, Mar 21, 2006 at 10:57:06PM +0100, Francesco Peeters (Asterisk)
wrote:
Why I have 'Format for call is ulaw'? I'd like to have alaw but keep
ulaw
to
accomodate errors in various configurations (if any, not here!).
EuroISDN uses uLaw
On Thu, March 16, 2006 12:08, Stojan Sljivic - GDS said:
Hi,
Does anyone know what would be acceptable RTT. Is 200ms OK?
Regards,
Stojan Sljivic
When any of my VPN tunnels get over 100ms I start to get worried! Avg
speeds on the tunnels are below 45 ms...
I guess it depends on the level
On Thu, March 16, 2006 22:38, rnacharya said:
Hi..,
we have two asterisk server interconnected to each other through IAX2
trunk in two separate office.
with this bellow configuration do we need to have Licensing for using G729
codec
Office A T1 - Astrisk
working though, so I guess that's something! G)
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate
satisfied with them and would recommend them to
others. (BTW: I do not sell SonicWALLs nor do I work for the company)
Other people like Zyxel's, which I think are crap... To each their own! ;-)
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't
Just in my Inbox:
Original Message
Subject: Over 40 destinations for FREE!
From:[EMAIL PROTECTED] [EMAIL PROTECTED]
Date:Thu, March 2, 2006 17:40
To:
--
Dear
On Thu, March 2, 2006 18:26, trixter aka Bret McDanel said:
On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote:
Just in my Inbox:
From the makers of Voipbuster: http://www.internetcalls.com
Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding!
Finerea
On Wed, February 15, 2006 22:35, Brent Torrenga said:
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
type of phone (79XX loaded with SIP firmware) as everyone else. He had
some
disconnects a few
On Tue, February 7, 2006 9:53, Sven Fischer said:
Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer:
Hello all,
if I try to call from one phone on the internal S0 to another on the
same
S0 using zaphfc, the bus is hung up. The called phone is ringing, but I
can't talk from one phone
On Tue, February 7, 2006 11:16, Peer Oliver Schmidt said:
Francesco Peeters (Asterisk) schrieb:
They have several ISDN BRI connections, most of which will be dropped.
Only one will be retained, for 2 reasons:
1) It has the ADSL link
2) The number has been the main contact number for over 20
I have a question,
I have to provide a solution for an office that will be almost abandoned,
and there will be one or sometimes two persons 2 days a week. The main
number however should be preserved.
They have several ISDN BRI connections, most of which will be dropped.
Only one will be
On Wed, February 1, 2006 22:12, Armin Schindler said:
On Wed, 1 Feb 2006, Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
chan_capi does not set the NT-mode. Your cards driver need to do that.
E.g. for Eicon DIVA Server cards, you just set the '-x' option with
divactrl
or
On Fri, February 3, 2006 0:44, Imran Ahmed said:
Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
The IVR records the conversation between the other partecipant to the
conference and
not find a way to do that
either.
Any suggestions?
What ISDN driver set are you using? (Zap/Bristuff/vISDN/mISDN/CAPI?)
I see (hear!) the same, but only when using vISDN, bot BRIstuff (haven't
tried mSIDN/CAPI)
--
Francesco Peeters
PalmOS user since Pilot1000
Tungsten|T3 owner, still
On Wed, February 1, 2006 12:07, Accursio Avona said:
Imran Ahmed wrote:
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know if IaxComm has inband dtmf mode available, if so enable
On Wed, February 1, 2006 15:04, Accursio Avona said:
Francesco Peeters (Asterisk) wrote:
SNIP
AFAIK there's no DTMF option in IAX2...
IAX always sends DTMF inline, eliminating the confusion often found with
SIP.
http://www.voip-info.org/wiki-IAX
If so, wy the IVR does not hear the dtmf
On Tue, January 31, 2006 14:35, bails said:
Hi all, Some friends of mine have an asterisk box which they use for
outgoing IAX2 via voipbuster.com.
They have been told that they now have an incoming number 0044117***
The thing is I cant seem to get any debug info on the incoming.
I have
On Tue, January 31, 2006 10:43, Juergen K. Zick said:
HI,
all newer HFC-S cards will do. Depending on your application and system,
you could easily ebaying an used Fritz!Card PCI or some active AVM B1
controller. Depending on the card you want to use you must se ZAPHFC or
mIISDN/chan_isdn or
On Fri, January 27, 2006 23:47, Script Head said:
What you're trying to accomplish can be easily done with an SQL query. You
need to create a table of all the prefixes (international dial+country
code+city/carrier) and join by that prefix.
On 1/27/06, Damon Estep [EMAIL PROTECTED] wrote:
On Fri, January 27, 2006 15:13, ram said:
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get
On Fri, January 27, 2006 16:09, Ian Cowley said:
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension
On Fri, January 27, 2006 17:23, Ian Cowley said:
Iax.conf
[general]
;bindport = 4569 ; Port to bind to (IAX is 4569)
bindport = 5036 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729 ; 4
On Tue, January 24, 2006 12:09, RumaTech said:
Hi, all
I have a problem using voipbuster (and voipstunt) for that matter.
On all calls, voice is disconnected after 30s. Asterisk still thinks that
call is in progress and I do not get any tones, just silience. Remote
party
gets normal tones
On Sun, January 22, 2006 13:02, Charles Wang said:
I have the same problem too.
I install the G.729 (IPP) to asterisk 1.0.x, and it works well.
When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine.
I can use show translation and find it too. But when I make a call
using
On Sun, January 22, 2006 19:40, Douglas Garstang said:
Hang on there's a non commercial G729 codec that will work with
Asterisk? Can someone point me to where I can find it?
Thanks,
Doug.
Intel provides a sample for non-commercial/testing.
http://www.voip-info.org/wiki-ITU+G.729
and
On Sun, January 22, 2006 22:32, Ron Wellsted said:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Guillermo Salas M wrote:
I've the same problem with sip1.sipdiscount.com. The calls are not
connecting but are billed.
SIPDiscount seem to have been having intermittent problems since Friday
I am trying to move from IAX2 to SIP for voipbuster, moving at the same
time to sip1.voipbuster.com.
When I try calling out, I see that there is SIP exchange, and in many
cases also RTP data being exchanged.
Hover in a very large number of attempts the connection is not
established. Half of the
On Sat, January 21, 2006 22:10, MapsAir said:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
I tried to follow the instruction from
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and
On Sat, January 21, 2006 23:21, Franz Bräuer said:
Hi,
MapsAir wrote:
Has anyone successfully Installing the none commercial intel g729 codecs
into [EMAIL PROTECTED] 2.2?
Installed them today. Installing from source didn't work for me (Debian,
Asterisk 1.2 from svn) but just adding the
On Fri, January 20, 2006 21:46, Roberto Pereyra said:
Hi
Someone have luck using Sipdiscount service with IAX ?
I only can use sipdiscount IAX service using a free account (only 1
minute
call) , I have a normal account and with it can login in the IAX server.
I using sip1.sipdiscount.com
On Thu, January 19, 2006 0:13, Hans Witvliet said:
On Wed, 2006-01-18 at 11:45 +, John Daragon wrote:
snip
You can't use a Digium card because Digium doesn't make an ISDN2 card.
snip
If i see how many questions/complaints there are on the list about
isdn/bri
i would allmost wonder
On Tue, January 17, 2006 22:10, Camilo Gonzalez-Cortes said:
The Fritz cards was not designed to run on asterisk whereas the following
German ISDN cards (http://www.junghanns.net/en/quadBRI_produkt.html) was
designed specially to run on this platform.
The only problem with this vendor is the
As I haven't received any posts since yesterday...
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
On Fri, January 13, 2006 8:51, Tomislav Parcina said:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Also: What are the SIP CanReinvite settings for these phones?
This shuldn't be important because he have w and W in his dial plan. *
doesn't allow reinvite if you have t, T, w or W.
On Fri, January 13, 2006 13:29, Tomislav Parcina said:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
It shouldn't make a difference, but should not and does not isn't always
the same thing!
We can't discus about this topic. It is simply meather of opinion. You
think that is
On Thu, January 12, 2006 19:18, Ben Ferguson said:
Hello all. I've been searching and can't quite find what I'm looking
for...
I've gotten AMP installed and up and running quite decently on an Asterisk
box and am now in the process of tweaking it to my needs. My company
currently has
On Fri, January 13, 2006 5:15, Jennifer Hales said:
Hello all,
I am unable to get automon recording to work; can someone advise me what I
am doing wrong? When I do *1 all I see in the CLI screen is attempting
native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call
record
On Wed, January 11, 2006 12:46, Tomislav Parcina said:
When I try to make attendend transfer (*2) this what hapends.
I press *2 other person goes on hold and I hear transfer. I press
extension number and that extension starts to ring but I don't hear
anything. If nobody picks up that phone
On Wed, January 11, 2006 7:52, scott said:
Hi All
Apologises if this has been disussed and I missed it.
My SetUp
I have a sip phone registered to an asterisk box (a1) in one location 1.
This phone dials an extension which is in another location, so a1 passes
the call via IAX to the other
On Wed, January 11, 2006 16:00, Colin Anderson said:
As a rule of thumb, I always explicitly set CallerID in my dialplan before
making a call through IAX, SIP or PSTN. If you make it part of a generic
dialout routine then it isn't a hassle. It always works.
It sometimes doesn't for my
On Wed, January 11, 2006 19:35, Stephen Bosch said:
I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers.
If I boot the machine without having the wcfxs module autoload, then
install the module with modprobe, asterisk works just fine.
If I boot the machine and autoload the
On Wed, January 11, 2006 21:36, Stephen Bosch said:
Francesco Peeters (Asterisk) wrote:
On Wed, January 11, 2006 19:35, Stephen Bosch said:
Try running ztcfg -vvv
Yes, that fixes it -- my question, I guess, is how to get that to run
automatically at boot time...
-s
Either put
On Wed, January 11, 2006 23:37, Tzafrir Cohen said:
On Wed, Jan 11, 2006 at 01:36:24PM -0700, Stephen Bosch wrote:
Francesco Peeters (Asterisk) wrote:
Try running ztcfg -vvv
Yes, that fixes it -- my question, I guess, is how to get that to run
automatically at boot time...
I run ztcfg
On Mon, January 9, 2006 16:44, Patrick Conroy said:
Does anyone know if it would cause problems to have the same Zap channel
in
multiple goups? So, for example, if I have two PRIs would the following
work or would it cause problems:
channel = 1-23
group = 1
channel = 25-47
group = 2
On Tue, January 10, 2006 6:03, Dovid B. Asterisk Users said:
Ken,
I would tell the client that you offerd phones for under $100.00 and he
didnt like them so now for a diffrent phone he will have to pay more. Also
I have an 841 and for it works great. I also installed one for a customer
in a
On Tue, January 10, 2006 5:50, Ira said:
At 05:44 PM 01/09/2006, you wrote:
We're getting our feet more and more wet with VOIP at work. We want
to experiment with a good wireless (as in WiFi) phone. What would
be a good phone to impress my boss with?
I have the Zyxel P2000W V2 and while it
On Fri, January 6, 2006 15:37, Michael Sampson said:
I work for a call center and we are looking at using asterisk to have
our operators take calls. Our message taking software records all the
calls on the operators computers. Right now we use these recording
controls from radio shack that
On Fri, January 6, 2006 15:46, Michael Sampson said:
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says so and so is on the phone for you, I say ok put
him through, she hangs up and I am
On Fri, January 6, 2006 20:20, Chandan Mishra said:
Hi
I have two asterisk servers. I just want to connect two asterisk server
using IAX2.
But the Asterisk Servers are not able to register each other. If some
body
have done this
then Please send me the configuration they have done in
1 - 100 of 256 matches
Mail list logo
