I am flying to Boston for a week, next week, and since there isn't
much in the way of places I can try SIP phones out in the UK, I was
wondering if anyone knew anywhere in Worcester / Massachusetts Area,
or just any US chain, what had IP phones available for viewing /
playing with.
Failing that I'
On Apr 7, 2005 5:50 PM, Paul Hewlett <[EMAIL PROTECTED]> wrote:
> On Thursday 07 April 2005 18:03, Wiley Siler wrote:
> > Go here and look at 'i'...
> >
> > http://www.voip-info.org/wiki-Asterisk+standard+extensions
> >
> > Thanks,
> > Wiely
> >
>
> Thanx. So I would change it to
>
> exten => _.,
On Apr 8, 2005 4:08 PM, John Novack <[EMAIL PROTECTED]> wrote:
> Bruno Hertz wrote:
>
> <>Jean-Michel Hiver <[EMAIL PROTECTED]> writes:
> Jean-Michel Hiver wrote:
> Oops, sorry for the list reply :/
>
>
> <>Actually, why does the Reply-To point to the Asterisk Users mailing
> list? This b
On 4/25/05, beonice <[EMAIL PROTECTED]> wrote:
> Hello, all.
>
> Google searches reveal that a couple of users have
> used Libretel port-of-call service via FWD using
> Asterisk. I don't have a FWD account. I don't have a
> SIP phone (no SIP setup at all, in fact).
>
> I just purchased a DID in B
On 6/12/05, Bob Goddard <[EMAIL PROTECTED]> wrote:
> > > You're fairly off base with that paragraph.
> >
> > you're fairly stupid.
>
> You are the one who is fairly stupid.
>
This thread is getting so OT and overall generally stupid its time it
died a gracious death, please... Granted there ar
Hi
I have been trying for a while to find a way to get an SMS send when I
receive a voicemail into my asterisk system. I don't want to send an
SMS if the caller doesn't leave a message. I have voicemail.conf set
up to email and delete.
302 => 302,Website Sales,[EMAIL PROTECTED],,attach=yes|dele
Hi
I have been lurking for a while, but now have a small problem or 3.
1) I have my inbound line via sip from VOIPUSER.ORG and have a simple
extension selection menu on my * box. Internally the DTMF tones are
present, (for xlite and * on same LAN), however calling in via the sip line
from a pstn
Hi
I believe this is due to the way US phone systems work, however I'm going to
ask anyway. In the UK there are several providers who provide national rate
PSTN => Voip gateways which are free to receive calls on, (for the
recipient), the caller pays the cost of calling. E.g 0844 0870 etc.
I a
Plus if you send your users to VoicemailMain(${CALLERIDNUM}) they don't hear
it at all.
They just get "enter password".
My 2c
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: 21 March 2005 16:05
To: Asterisk Users Mailing List -Non-
Hi all
I have been googling and wiki-ing and have found a number of potential
solutions to my questions, but I don't want to have to play about for too
long and risk messing up my * box now I've just got it working, if one of
you kind folk could offer your 2 penneth, (being a Brit I'll have none o
Hi
I tried to setup this system, and tried to follow their, bayhamsystems,
instructions for installing the app version of the util, but there was no
makefile.patch in the apps source folder so that failed, (I don't really
understand the make business on linux anyway), so I tried to use the perl
AG
7;t work.
I'm a newbie but I think that is the problem.
JD
Mark Charlton wrote:
>Hi
>
>I tried to setup this system, and tried to follow their, bayhamsystems,
>instructions for installing the app version of the util, but there was
>no makefile.patch in the apps source fol
on how to
upgrade my * I would be most greatful. I assume the * build in [EMAIL
PROTECTED] is a
standard distribution, which I can just upgrade somehow.
Thanks again for any help.
Mark Charlton
___
Asterisk-Users mailing list
Asterisk-Users@lists.
linked in, and then probably a cheap SIP / IAX hardphone.
Cheers
Mark Charlton
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean collins
Sent: 30 March 2005 00:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-U
a
beginner's toy.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Charlton
Sent: Wednesday, March 30, 2005 11:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Upgrade [E
I registered 1 week ago, and this message took 3 minutes to reach me.
Granted I'm in the UK so there is bound to be some strange effect causing
the speeding up of the message.
This topic is like a bad penny, it just won't go away.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mail
On Apr 6, 2005 2:29 PM, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is there some sort of Syntax checker for the Asterisk config files. Also
> is the /var/log/messages the only file that Asterisk dumps messages too
> or is there another place. My system works but I want to make sure I am
> not
On 6/30/05, Mike Dent <[EMAIL PROTECTED]> wrote:
>
> Here is what mine looks like:-
>
> exten => h,103,DeadAGI(fastsms|44797149|Uberetchs Voicemail.
> Caller ${CALLERID} left a new voice mail for sales a
> t ${DATETIME}|441524342XX)
> Hope that helps,
> Mike
>
I started out using the solution
On 7/1/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:
> > I have been trying for a while to find a way to get an SMS send when I
> > receive a voicemail into my asterisk system. I don't want to send an
> > SMS if the caller doesn't leave a message. I have voicemail.conf set
> > up to email and del
On 7/1/05, Peter Bowyer <[EMAIL PROTECTED]> wrote:
> On 01/07/05, Mark Charlton <[EMAIL PROTECTED]> wrote:
> > >
> > I have been fighting with the Bayham Systems FastSMS AGI script, and I
> > re-wrote it as a stand alone Perl script. I am now calling it wit
On 7/2/05, Anton Krall <[EMAIL PROTECTED]> wrote:
> Guys.
>
> Anybody know why sometimes on some servers Asterisk shows more than once
> while doing a ps?
>
> [EMAIL PROTECTED] akrall]# ps -ax|grep asterisk
> 20555 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk
> 20557 ?S 0:00
I have googled this to death, and all I get are reference to the MoH
needing a Zaptel timing source, and then people saying no they don't
any more.
-- Set Response Timeout to 2
-- Executing BackGround("SIP/211-57ba", "my-greeting") in new stack
-- Playing 'my-greeting' (language 'en')
On 7/3/05, Jerry Glomph Black <[EMAIL PROTECTED]> wrote:
> On Sun, 3 Jul 2005, chawki hammoud wrote:
>
> > Hi:
> >
> > I have my Asterisk server behind a nat and I want to
> > buy a static IP. Is there a company that sell IP and
> > forward it to IAX file as in the DID service. Any
> > reference o
On 7/4/05, Ronald_Wiplinger <[EMAIL PROTECTED]> wrote:
> Robert Goodyear wrote:
>
> >> I am confused about one of my installed server
> >>
> >> The dial plan seems to be ok, but sometimes NOTHING happens if I
> >> try to dial an extension (from X-Lite), next time it is done.
>
On 7/1/05, Mark Charlton <[EMAIL PROTECTED]> wrote:
> On 7/1/05, Wilson Pickett <[EMAIL PROTECTED]> wrote:
> > > I have been trying for a while to find a way to get an SMS send when I
> > > receive a voicemail into my asterisk system. I don't want to send an
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