On Mon, Feb 09, 2004 at 02:29:50PM -0500, Jess Magnaye wrote:
> have you tried this gs-102 with pppoe? verizon dsl uses
> pppoe. pppoe is
No, I didn't try. Yes, pppoe is fairly standard DSL stuff (when used
with an ethernet modem).
> logically like dhcp, but using ppp for added feature like aaa
On Mon, Feb 09, 2004 at 11:21:42PM +0100, Tomas Prybil wrote:
>
> How would you "roll out" a SIP based VoIP platform to to endusers with
> various connection solutions. Is there such a thing that solves the
> various issues of NATting a phone?
>
Well, there is not "one-fit-all" solution. GS ph
On Mon, Feb 09, 2004 at 05:37:48PM -, David J Carter wrote:
> Have a look at http://www.plugndial.com/aps_sample.html
>
I've been told by sipphone that this format is "new". It's not
supported by anything on the market right now.
--
Nicolas Bougues
Axialys Interactive
__
Contact me off-list and I'd like to login to your machine and try a fix on
it. Thanks!
Mark
On Mon, 9 Feb 2004, Michael Welter wrote:
> Today I had five system freezes. After re-examining the log file I see
> the following line precedes each freeze:
>
> "Got event 2 (Ring/Answered)"
>
> Howeve
Paul Mahler wrote:
What are you trying to accomplish? What is the architecture of the system
you are trying to get operational with NAT?
How would one deploy users with SIP phones behind various kind of
routers/firewalls?
The assumption must be that the users not will be able to handle
conf
Thanks! I'll try this one. This is not standard part of Asterisk bundle,
as chan_skinny is? Am I right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vic Cross
Sent: Tuesday, February 10, 2004 12:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
Do you pay the G729 licences fee?
- Original Message -
From: "Anthony Law" <[EMAIL PROTECTED]>
To: "Mailing List Asterisk" <[EMAIL PROTECTED]>
Sent: Monday, February 09, 2004 6:00 PM
Subject: [Asterisk-Users] Re: question for oh323 users
> Thanks very much Michael.
>
> It worked but onl
On Tue, 10 Feb 2004, Tomica Crnek wrote:
> Thanks! I'll try this one. This is not standard part of Asterisk bundle,
> as chan_skinny is? Am I right?
That's right. As I mentioned, I don't know if there is an intention to
bring it in (Jeremy has said in the past not to expect too much of
chan_s
As I said in my first post, I don't think this is an asterisk problem.
Last night I swapped-out the Netgear NIC card, and things appear to have
settled down (no more event 2 messages).
I really appreciate all the help I received from the list members and Mark.
I'll update tracking bug 963 when
greetings
i have installed Asterisk on OSX 10.2.8
i have two xlite phones.
'i can connect to the PBX and access the menu,but i don't know how i
would connect to the other phones.
When i dial 1234 i get the operator, so i am making a connection. Where
do i add the users? is there something more i
It is important to note that cheap nic cards that were really designed for
1% utilization on a workstation are not well suited for an Asterisk server
installation with any kind of VOIP traffic. We foolishly put a Realtek card
in a test server and after a month literally fried the NIC card, It was
e
I'm having some problems with a SIP FXO gateway working with Asterisk when a
call that involves the gateway is put on hold. This gateway was working
up to a firmware
upgrade but I believe it may have been working for the "wrong reasons".
Here is what
happens:
1. User calls in from PSTN to SIP F
Hm. After seeing all the people who say it works, I thought - maybe I
forgot to dial 9 in front of the number and that's why the call failed.
So I looked up the Wells Fargo toll free number again and tried it.
Failed. SIT tones and "We're sorry, your call did not go through. Will
you please try
I have one issue - spurious inband DTMF detection results in the caller
hearing the odd burst of DTMF at random. My calls are going from a
Cisco ATA 186 running Version: v2.16.1 ata18x (Build 030709a) using the
g.711 codec with the AudioMode parameter set to 0x11241124, to a
default-installed Aste
There are any way to connect an ATA-186 to an legacy PBX using the fxs port ?
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Are there any way to connect an ATA-186 to a legacy PBX using the fxs port ?
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On Tue, 10 Feb 2004, Tim Petlock wrote:
> Hm. After seeing all the people who say it works, I thought - maybe I
> forgot to dial 9 in front of the number and that's why the call failed.
>
> So I looked up the Wells Fargo toll free number again and tried it.
> Failed. SIT tones and "We're sorry,
Hi,
I'm testing outbound calls for the fist time, using isdn4linux and a cheap
20$ ISDN CARD: it works !
I have more problems restricting pstn calls can I allow inbound sip
access to ALL asterisk features ONLY from the requests sent by my Ser
proxy/registrar ?
In Ser I use this to rewri
I am planning to deploy a fairly standard PSTN-dodging setup, like this:
T1 - * - IAX2 - * - T1
In other words, two Asterisk boxes with T100Ps, interconnected with IAX2,
used for bridging calls from one PSTN endpoint to the other. What makes
this interesting is that on the receiving end, there i
Wes Marderness wrote:
What does your extensions.conf look like? Did you answer() the call first ?
The relevent sections of extensions.conf:
[voicemail access]
;Extension 8 to get to voicmail:
exten => 8,1,Answer
exten => 8,2,VoicemailMain
[wellingborough-road]
;includes
include => emergency
inclu
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At 1:53 AM -0500 2/10/04, James H. Cloos Jr. wrote:
"Kris" == Kris Stark <[EMAIL PROTECTED]> writes:
Kris> On a different note - is something up with the freenum.org enum
Kris> lookups? ... I've had them fail on all US numbers...
The nameservers for freenum.org. have glue records for 1.freenum.or
Yes, you need G.729 codec.
Michael.
Anthony Law wrote:
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to tr
Feb 10 09:57:37 WARNING[98311]: db.c:46 dbinit: Unable to open Asterisk database
I'm seeing this in my logs, 0.7.2 (debian package). I looked through the
source, and thought it was looking for a RDBMS (mysql/postresql), so
I set both of them up (one at a time), and I'm still getting it.
What is th
Hi All!
I have this problem with callerid detection with my
x100p here in brazil., my line have this function and it works with a very cheap
aplliance that i have here in the office, here in brazil it is called
"detecta".
I think that the caller id info comes in DTMF
before the 2 ring of
Last 2 days I have noticed that more and more often calls are just being
dropped. I can't find any logs or anything indicating that something is
wrong. If I do a trace and wait for a call to drop I can only see hangup
and nothing else. Sometimes calls do last for minutes without problem
and someti
Hi!
> Feb 10 09:57:37 WARNING[98311]: db.c:46 dbinit: Unable to open Asterisk database
>
> I'm seeing this in my logs, 0.7.2 (debian package). I looked through the
> source, and thought it was looking for a RDBMS (mysql/postresql), so
> I set both of them up (one at a time), and I'm still getting
Sigh. As soon as I send e-mail, I find the answer. /var/lib/asterisk/astdb
is the file. Was owned by root, not asterisk (at least on Debian).
Time to file a package bug report for Debian.
Tim
--
><
>> Tim Sailer
On Mon, 2004-02-09 at 13:56, WipeOut wrote:
> Steve Kennedy wrote:
> >Probably a dumb question, but what's the best Linux variant to use to
> >build/run an Asterisk server.
> >
> >Hardware is Compaq DL360 with a Widcard 410.
> >
> >Debian/Fedora Core ?
> >
> Which ever one you are most happy with i
Bisker, Scott (7805) wrote:
> Has anyone experienced problems with dialup through asterisk. I'm
> having some varied success with dial-in and dial-out.
>
> All my analog extensions are connected to * via Adtran 750 FXS
> channelbanks using FXO_KS signalling. I have a longdistance T-1
> (e&m_w) fr
I have a sip box with two FXS ports (Draytek 2600v adsl router)
I have had very little luck getting the two talking together.
For a very short time I did have calls originating on my FXO card routed
to the phone working.
Phone1/2 on router ---> handytone works
handytone ---> r
You
are right, Brazil uses DTMF caller ID.
The
format is very simple
Asterisk has all the tools available to get DTMF caller
ID to work. (DTMF decoder routines,etc.) and T1-CAS uses a very similar
format.
I
guess somebody just needs to spend the time and programm it into the zaptel
d
CW_ASN - Gus wrote:
You must register with cisco in order to get ata image.
I tried, but Cisco (Germany) has no idea how to do this...
BTW, my ATAs sometimes cannot make calls. I first have to make a call to
one "ATA-Extension", wait for the Phone to ring, then i can make calls
again.
I am
Plug the ATA Fxs port into your PBX FXO port.
On Tue, 2004-02-10 at 07:55, Miguel wrote:
> Are there any way to connect an ATA-186 to a legacy PBX using the fxs port ?
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.co
Ok!
I hope some *guru can make it soon... :-)
but i´m happy to know that my guess is
correct!
thank´s
Miklos
- Original Message -
From:
Alfred R.
Nurnberger
To: [EMAIL PROTECTED]
Sent: Tuesday, February 10, 2004 12:48
PM
Subject: RE: [Asterisk-Users] Ca
I have this problem intermittently, and doing an asterisk
-r showed "too many retries." hunting around with
ethereal found a bad hub.
-e
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
Of
> Tomica Crnek
> Sent: Tuesday, February 10, 2004 9:23
On Tue, 2004-02-10 at 07:01, mattf wrote:
> It is important to note that cheap nic cards that were really designed for
> 1% utilization on a workstation are not well suited for an Asterisk server
> installation with any kind of VOIP traffic. We foolishly put a Realtek card
> in a test server and af
I think the "please try your call again later" part of
the recording is to be taken seriously.
I've tried 800 numbers and failed but I tried re-dial several
times and it worked after a few attempts. I have no idea how
my call is routed but I'd not be supprized if NuFone, Iconnect
FWD and the lik
On Tue, 2004-02-10 at 08:53, Mark Johnston wrote:
> I am planning to deploy a fairly standard PSTN-dodging setup, like this:
>
> T1 - * - IAX2 - * - T1
>
> In other words, two Asterisk boxes with T100Ps, interconnected with IAX2,
> used for bridging calls from one PSTN endpoint to the other. Wha
"David J Carter" <[EMAIL PROTECTED]> wrote:
>Matteo,
>
>try: -
>[incoming]
>include => default ;default location for internal phones
>exten => s,1,Answer
>exten => s,2,Wait 10
>exten => s,3,Dial(SIP/100)
>exten => s,4,Hangup
I don't think that will work. There is no mention in the
documentat
Bill Michaelson <[EMAIL PROTECTED]> wrote:
>I am trying to muddle my way tthrough getting something - actually
>anything to work - with Asterisk. I've acquired a Grandstream phone and
>I've got * on a Red Hat 9 box. I've gotten to a point where I can see
>(via ethereal) that the phone REGIST
VOIP is a very low data rate compared to the bandwidth of
a switched 100BaseT network. Lets say you are using 100BaseT
to trunk 100 simultainous calls at 64Kbps (How many of us really
would ever do that?) 100 calls would be 6,400,000 bps. Well
over what a T1 could handle but is only 6.4% of 100
Hello to everybody!
I have mounted an asterisk pbx on my work, which have two nic cards,
one with the Public IP onto the DMZ and the other with the Private IP to
connect the LAN network, and one sip phone (grandstream BT100) on internet
in a DSL. The LAN side works perfectly but the inter
> I don't have a Nufone account (Jeremy - if you are reading - I would
> probably have one if there was a price for a starter package listed on
> your site - something for SoHo use, without any deep discounts or
> anything, just something to use to play with the service; I have a
> personal aversi
Here's a quote on Realtek 8139 NIC cards from the comments in the FreeBSD
kernel:
"The RealTek 8139 PCI NIC redefines the meaning of 'low end.' This is
probably the worst PCI ethernet controller ever made, with the possible
exception of the FEAST chip made by SMC. The 8139 supports bus-master
DMA,
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Greg Boehnlein
> Sent: Monday, February 09, 2004 10:50 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Dialing 800 numbers with VOIP
>
>
> On Mon, 9 Feb 2004, Tim Petlock wrote:
>
[...]
I second the last comment below. I sent $5.00 via pay pal
to NuFone and was up and runing very quickly.
My problem is billing. What you do is send e-mail asking
"How much is left in the account?" and then if it is "low"
you send more. For test purposes this is not bad but I
can't imagine set
That is a nice thought and takes care of problem with fax transmission over
IP. However not applicable in my case though. Most 'fax users' will not have
Asterisk on site but some other fxs gateways instead. Also from what I've
read RxFax tends to cause problems.
Best regards,
-Original Messa
On Tue, Feb 10, 2004 at 10:36:45AM -0800, Chris Albertson wrote:
> Why is this not on the web and automated? My guess is either a lack of
> devlopment resources at NuFone or he's more interested in supporting
> high volume customers who don't need a web site but more likely both.
Apparently, you
Thanks for the email William.
I guess the main challenge is to setup the system in a way that's
manageable. I didn't really understand your voicemail notification idea. So
when vmail is left at the central server, you have the server call the
remote office extension and leave a vmail there that th
Hi,
I bought a TDM developers kit and have installed teh TDM400 with one
port in my system. The card is showing up as a Tiger Jet network
interface and is not found by the zaptel drivers.
I am running RH9 with all the latest updates. Any help for getting the
card to work is appreciated.
I looked
I am having a problem with the logger stopping at random points. It will
then work again randomly, and then stop again.
For example I know warnings have popped up, but they do not go in the
messages file. I see them occur on the console and they are never written
to file. I then notice logs lik
It shows that way on my RH9 box, but that is not what's causing your
problems with the drivers not seeing the card. Have you configured your
zaptel.conf for your hardware? Have you done ztcfg -vv? What order did
you modprobe? We need a little more info to help Thanks!
Sean
-Original
Rob Fugina <[EMAIL PROTECTED]> wrote:
>Is this really a company, or is NuFone in some guy's basement?
Maybe both. :)
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com
___
Hi,
I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High traffic.
Thanks in advance,
Daniel
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Might be, but even if you are not using voip, calls drop. I have a 2 E1
links and bridged calls between them drop from time to time.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ejay Hire
Sent: Tuesday, February 10, 2004 5:46 PM
To: [EMAIL PROTECTED]
S
Daniel Bichara wrote:
Hi,
I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High
traffic.
the only one you'll get on the phone there is Fidel Castro (which is the
only one to have internet access too)
:D
Amaury
Thanks in advance,
Daniel
_
That sounds like a classic issue of busydetect=yes and callprogress=yes
in zapata.conf. Don't do that. Set them to no
On Tue, 2004-02-10 at 14:16, Tomica Crnek wrote:
> Might be, but even if you are not using voip, calls drop. I have a 2 E1
> links and bridged calls between them drop from time t
Hi,
I'm generating an error in the logs that looks like this:
Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the past?!?!
This error is triggered when I execute the following line as part of my auto
attendant.
exten => s,2,Wait,1
Any suggestions are fantastically appreciated.
_
On Feb 10, 2004, at 9:21 AM, Chris Albertson wrote:
(Ethernet does not work well when loaded over 30% of
it's nominal bandwidth.)
It's a common myth, but it's never really been true, even in the days
of big shared Ethernet segments. On a modern switched network, you
should really be able to get
Amaury Jacquot wrote:
Daniel Bichara wrote:
Hi,
I am looking for a VoIP (SIP or Asterisk) termination at Cuba. High
traffic.
the only one you'll get on the phone there is Fidel Castro (which is
the only one to have internet access too)
:D
You are right! But someone must have a rate better
Daniel
Well, that seems like a good joke, but forget it!. Honestly, and don't
take me wrong. I am cuban and it is one of risky and hard to maintain
deals in the whole worldwide telco business.
Is better to get termination in the Polinesian Island than Cuba
But, if you get it let me know I will
Hi,
I had similar problems. I removed the card from the system and ran kudzu, it
prompted to remove the tiger network card; i removed it. Then again
installed the card in the system and ignored it in kudzu screen, i was able
to load the zaptel drivers.
Give it a try.
Deepak
- Original Messag
Thanks for all the ideas!I looked around on the internet last
night to try and find some ideas for the wiring and came across some
sites that have started me thinking about how the phone might be wired to
get the intercom to work. I thought I would pass what I found on
to everyone in case they foun
Okay,
So here is a another relatively neophyte question that I hope not to get
RTFM flames for. I am curious if there is a way that I can use incoming
caller id and pass it to a an application so that a CSR can get the
information of the customer that is calling without having to ask the
custo
On Tuesday 10 February 2004 14:36, Christopher J. Wolff wrote:
> Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the
> past?!?!
Your machine is heavily loaded. A thread which was interrupted to
schedule another thread or process was not able to complete its task
in time. Your solution i
Hi Sergio,
I don´t have any setup like you but looking over you config I saw this:
> My capi.conf is the next:
>
> [global]
> mode=immediate
> isdnmode=multipoint
>
> txgain=0.5
> rxgain=0.5
>
> [interfaces]
> msn=951014943
> incomingmsn=951014943
> controller=1
> context=default
> echocancel=
Hello Daniel,
What is your target price to Cuba? We can provide A to Z termination for
Asterisk SIP to any destination.
Regards,
Aram Ter-Martirosyan
Senior Account Manager
Hi-Tech Gateway, Inc.
http://www.hi-teck.com
1225 Grand Central Ave.
Glendale, CA 91201
[EMAIL PROT
On Tue, 2004-02-10 at 15:02, Yonah Wolf wrote:
> Okay,
>
> So here is a another relatively neophyte question that I hope not to get
> RTFM flames for. I am curious if there is a way that I can use incoming
> caller id and pass it to a an application so that a CSR can get the
> information of th
Hi all,
I will ptobe your answers tomorrow. I'll say the results.
Thanks for all.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Sascha
Knific
Enviado el: martes, 10 de febrero de 2004 22:08
Para: [EMAIL PROTECTED]
Asunto: AW: [Asterisk-Users] Eico
It's noteworthy that while Linux is GPL'ed, that doesn't mean that the
userspace applications that essentially make up the Snom phone and run
on top of the GPL'ed Linux kernel are.
Snom will gladly give you their customizations to the kernel, and a
build environment that will produce a firmware im
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina
> Sent: Tuesday, February 10, 2004 2:28 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP
>
> Is this really a company, or is NuFone in some guy's basem
Darren,
To achieve voicemail on a central location I did the follow.
In a context I call
exten => _[1-5]XX,1,Macro(stdexten,${EXTEN})
So extensions 100-500 are all routed thru a macro unless previously
defined
macro-stdexten contains:
[macro-stdexten]
exten => s,1,DBget(caller=EXTEN/${A
Title: RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk
Well, I've got my Vega 50 Analog able to pick any available analog port for an outgoing line.
Let's assume we've started with the QuickStart instructions that came with the Vega:
- First, add a new Planner Group. Give it a name (
I'll second this.
For the past 4 days, Vonage can't figure out how to process our visa check
card. In the meantime, Nufone has us setup with an account, ready to roll.
- Chris Clifton
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 10, 2004
On Tue, 10 Feb 2004, Matthew Hardeman wrote:
> It's noteworthy that while Linux is GPL'ed, that doesn't mean that the
> userspace applications that essentially make up the Snom phone and run
> on top of the GPL'ed Linux kernel are.
>
> Snom will gladly give you their customizations to the kernel,
yOn Tue, 10 Feb 2004, Tilghman Lesher wrote:
> On Tuesday 10 February 2004 14:36, Christopher J. Wolff wrote:
> > Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the
> > past?!?!
>
> Your machine is heavily loaded. A thread which was interrupted to
> schedule another thread or process w
There was an article in El Pais' (Uruguay) Sunday paper two weeks ago
about internet access in Cuba. Tourists can buy cards (priced in US
dollars) that allow dialup access. According to the article, the call
to the dialup number can only be placed from phone accounts that are
billed and paid in U
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ringtones on the phone (in the section "Support for SIP Alert-Info
Header"), but I can't get anywhere with it.
"...the Alert-Info header consists of a name of an
Sorry, we have to make some money... Product business is tough!
:-) CS
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Greg Boehnlein
> Sent: Tuesday, February 10, 2004 10:39 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users]
In RFC-3261, the Alert-Info header is specified as a URL. When the
Alert-Info header is received, the phone downloads the file from the URL
and plays it as the alternate ring tone. This release does not support any
external ringers. Only the tones and ring patterns that are already
internal to the
BTW if you want to use Alert-Info on snom just provide an http uri which
points to an 8 kHz mono 16-bit sample WAV file.
CS
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Creel
> Sent: Tuesday, February 10, 2004 11:08 PM
> To:
After having this bother me for a while and trying to do it
via the source I finally got * to give me a different dial tone for internal calls
versus an outside PSTN call.
I have included the dial plan from extensions.conf.
Background info:
Inside is the context defined in my Za
>I am trying to muddle my way tthrough getting something - actually
>anything to work - with Asterisk. I've acquired a Grandstream phone and
>I've got * on a Red Hat 9 box. I've gotten to a point where I can see
>(via ethereal) that the phone REGISTER's successfully
IMHO snom phones are ugly. Now this is my view but not
everyone will agree with me. Not ment to start a "My Phone is better than
your phone"
bkw
On Tue, 10 Feb 2004, Christian Stredicke wrote:
> BTW if you want to use Alert-Info on snom just provide an http uri which
> points to an
If its all zap what was wrong with ignorepat? does it not do this ? I may
be wrong.
bkw
On Tue, 10 Feb 2004, Alex Lopez wrote:
> After having this bother me for a while and trying to do it via the
> source I finally got * to give me a different dial tone for internal
> calls versus an outside P
Christian,
Where is a good place to purchase your phones in Germany? I found a
distributor in the UK but maybe just am not looking in the right place for
Germany.
Thanks,
Robert
American Expatriate in Friedrichshafen (Grund oder Entschuldigung für die
englisch)
Christian Stredicke said:
> Sorry,
Bill Michaelson wrote:
I am trying to muddle my way tthrough getting something - actually
anything to work - with Asterisk. I've acquired a Grandstream phone and
I've got * on a Red Hat 9 box. I've gotten to a point where I can see
(via ethereal) that the phone REGISTER's successfully with
Hiya,
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ringtones on the phone (in the section "Support for SIP Alert-Info
Header"), but I can't get anywhere with it.
the only thing i'm getting, is using
exten =
Good point Brian, but ignorepat only gives you the SAME dial tone not a
different one!!!
I would love to see an option for ignorepat that would do this!!!
Say ignorepat => 9,Playtone(350+440)
Alex
Date: Tue, 10 Feb 2004 16:36:05 -0600 (CST)
From: Brian West <[EMAIL PROTECTED]>
To: [EMAIL PROTEC
Does anyone know how to make the 7960 “messages”
key dial voicemail? SIP 6.0.
Thanks!
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
exten => 555,1,SetVar()
Will do what you want.
Andreas Anderson wrote:
Hiya,
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ringtones on the phone (in the section "Support for SIP Alert-Info
Header"), but I
This will give you what you want.I type a little to fast for
the brain buffer sometimes.
exten => 555,1,SetVar(ALERT_INFO=)
Andreas Anderson wrote:
Hiya,
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ri
In SIPDefault.cnf, set the following:
messages_uri: "76"
In extensions.conf, have a line that looks like this:
exten => 76,1,VoiceMailMain2([EMAIL PROTECTED])
John
- Original Message -
From: Paul Mahler
To: [EMAIL PROTECTED]
Sent: Tuesday, February 10, 2004 6:01 PM
Subject: [Asterisk
I signed up with these guys on a whim. $20 a month for unlimited US/Canada
local and long. You can't take your numbe over there but they'll give you
a new one.
So to my problem; how do I set them up as a SIP peer. I have a stanza in
my sip.conf as follows
register => 9733878280:username:[EMAIL PR
On Tue, Feb 10, 2004 at 03:04:41PM -0600, Tilghman Lesher wrote:
> On Tuesday 10 February 2004 14:36, Christopher J. Wolff wrote:
> > Feb 10 13:19:36 NOTICE[17743913]: Request to schedule in the
> > past?!?!
>
> Your machine is heavily loaded. A thread which was interrupted to
> schedule another
Just recently my calls through iaxtel to FWD user do not go throuhg due
to busy circuit.
Wonder if there is any change to the setup of Iaxtel?
-- Executing Dial("SIP/1002-246A",
"iax2/xxs:[EMAIL PROTECTED]/[EMAIL PROTECTED]") in new stack
-- Called xxx:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call
On Wed, Feb 11, 2004 at 02:02:03PM +0100, dkwok wrote:
>
> -- Format for call is G729A
^
I suspect that if you use a standard
format your call will go through. Also
keep in mind that there is no reason
to go through IAXTel for this -- it is
just necessary to dial SIP/[E
Hi Everyone,
I just having my first expierence with Asterisk and after solving the first
little
problem now I am stuck a little. Perhaps anyone can help.
Running Debian/Woody w/ 2.4.18 kernel ... think I have installed all necessary
packages for running Asterisk.
I downloaded the CAPI driver for
We will be install a 4xE1 card, but, I need to order the
system first. How powerful of a machine will we need to purchase to make sure I
can fully utilize the card and not create a bottleneck with the system (e.g.
processor, memory, etc…)
dank
Dan Kordick
BlastSpot
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