I resolved this - I'm in the UK, and the problem was due to the cable (you need a two wire RJ cable) - I replaced it and it worked fine.
Thanks
On 7/13/05, Luki <[EMAIL PROTECTED]> wrote:
John,all this "ringing" makes me think that your "PSTN Ring Timeout" is too
low. Increase it by a second or
My porblem is incoing PSTN calls are being forwarde to the * box, the
phone rings, but when the phone is picked up, the call is not taken -
it continues to ring.
I am forwarding the call to () in my dial plan
Can anyone assist? This is driving my crazy!
Extract from the * console
Exe
Has anyone any experience of the above.
Key feature for me is tracking incoming and outgoing emails and
linking them to the contact record.
Thanks, sorry for the OT ;-)
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On Fri, 04 Feb 2005 17:26:22 -0800, Steven P. Donegan <[EMAIL PROTECTED]> wrote:
> Is there any support in Asterisk for encryption of IAX and/or any other
> VOIP protocols? I haven't seen anything on this in the wik
Hi,
I've got Outlook to call the number on * using the TAPI interface
documented on the Wiki. Its working OK.
I have downloaded the Indentapop application, and it appears to
connect to * Ok using the Debug modes, but It isnt detecting incoming
calls.
Has anyone git identapop working?
Care to sha
Hi,
Has anyone got incoming IAX to work on the above router.
I can call out, but incoming calls are not reaching the * box.
Has anyone got this working? Could they give me some configuration hints.
Thanks
John
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Anyone got any experiences of these with *, and also costings?
Thanks
John
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www.signate.co.uk
There is an e-book version.
I bought mine from the states, arrived very quickly to the UK - around
5 days, and no postage cost.
I ordered the CD of Asterisk with it, but didnt use it, and dont see
it as having much value.
Book is quite good for getting * running from basics IM
Hi, I've looked at the Wiki for this, have seen the Swift.agi details,
but has anyone got a current script for Cepstral and an example of
integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
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Has anyone used one of these with *, any observations/comments please?
Thanks
John
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Steve, thanks for that post, very useful and constructive.
Thanks ;-)
>
> >
> Check out the following link - proclaims to work with Asterisk
>
> http://www.sangoma.com/products/p_voice-data.htm
>
> --
>
> "They that give up essential liberty to obtain temporary safety,
> deserve neither libert
Now that I have your attention ;-)
Anyone know if a new release is planned, and if so when?
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What firmware version are you using now?
On Thu, 13 Jan 2005 12:38:09 -0600, Brian K. Hershey
<[EMAIL PROTECTED]> wrote:
> Paul-
>
> I have a ZyXel Prestige 2000W. It seemed to be junk at first,
> (not saving settings, flaky network connectivity, etc.)
> But upgrading it to the latest firmware s
When you say CVS HEAD is the the same as stable? where do you get it
from and what params do you use?
On Wed, 12 Jan 2005 17:03:34 +, Niksa Baldun <[EMAIL PROTECTED]> wrote:
> There is no easy answer to your question. If you ask me, I prefer not to use
> any patches, except that I am forced t
I use www.piebox.com (a Redhat AS Clone) which provides a good
compromise between stability/release testing and cost - however I'm
just using it on a test machine.
John
On Wed, 12 Jan 2005 16:58:29 +1300, Imran Sadiq <[EMAIL PROTECTED]> wrote:
>
>
> Could anyone please advise me on the best fl
Isn't this used as a timer source by zaptel?
On Wed, 12 Jan 2005 00:14:30 +0200, Shoval Tomer <[EMAIL PROTECTED]> wrote:
> You can disable the USB in the BIOS of the machine if you don't plan on
> using it.
>
> > -Original Message-
> > From: Michael Welter [mailto:[EMAIL PROTECTED]
> > Se
Not an enterprise level system, but anyone used the www.intertex.se IX66?
On Mon, 10 Jan 2005 10:14:46 -, Craig Waddington <[EMAIL PROTECTED]> wrote:
>
>
> We are on the lookout for a Firewall which is SIP aware, to pass the voice
> stream to Asterisk.
>
>
>
> We have looked at the Inga
Hi,
If I need to connect a home based user to an Asterisk server, how does
the above work?
Is it (after being configured/provisioned) plug and play?
Anyone done this got any comments
Thanks
John
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Anyone know in the current zaptel drivers and stable asterisk what the
parameters are to receive caller ID in the UK over BT lines?
Thanks
Looked at the Wiki and bugs.digium but more confused, perhaps someone
can help me
John
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I am tying to clear down an asterisk source directory before CVS'ing a
new version
the --ignore... option is being used but its still not being deleted,
can anyone give me some clues.
Sorry I'm new to Linux, as if you havent guessed. Googling hasnt helped so far
Thanks
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Anyone help me, I've looked at the Wiki and cant see anything
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I've an issue with my TDM4000P card and I will be calling Digium later
to ask for their help.
Could anyone help me with a basic configuration so they can SSH to me?
Thanks
John
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nge state message).
> I am sure the call progress and busy detect are both no in my conf. I
> am looking for a "More Correct"
> answer to this as well.
>
> On Jan 6, 2005, at 4:34 PM, John Middleton wrote:
>
> > What do you mean, as your first priority, you mean exten
nd callprogress cause my X100P's to not answer calls if I
> have them enabled.
>
> Phil.
>
> On 6 Jan 2005, at 22:56, John Middleton wrote:
>
> > what should those two settings say? should i set them to yes, or take
> > the lines out?
>
> >>
Yeah, it is set to the right signalling..
On Thu, 6 Jan 2005 22:47:45 + (GMT), Chris Glover
<[EMAIL PROTECTED]> wrote:
> On Thu, 6 Jan 2005, John Middleton wrote:
>
> > Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine,
> > zttest comes back with
Yeah, it executes the answer in the script, and goes on to play music
etc, but the line isnt actually answred IE continues ringing
On Thu, 6 Jan 2005 22:41:09 +, Phil Quinney <[EMAIL PROTECTED]> wrote:
> Hi John,
>
> Have you got a line like this:
>
> Exten => s,1,Answer
>
> You need to ac
Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine,
zttest comes back with configured. If i call a line when zttest it
shows on the display,and then goes when the line drops.
In * when a call comes in, it follows my dialplan and answers the call
according to the log, but IT DOESN
> [EMAIL PROTECTED] mohmp3]#
>
> Steve
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > John Middleton
> > Sent: Wednesday, January 05, 2005 2:06 PM
> > To: asterisk-users@lists.digium.c
Hi
On the www.asterisk.org main page it says "Music provided by Freeplay
Music" with a link - Where is the music in the *config? I cant find
any supplied music - is there any?
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See
http://www.wheely-bin.co.uk/asterisk/ check this link - I've
implemented it and it works, at least in the test environment.
John
On Wed, 5 Jan 2005 16:00:56 +, Mike Dent <[EMAIL PROTECTED]> wrote:
> Hi,
> Is there some script which can be called from a * extension to
> playback the rec
Could you please explain or tell me where it is explained the version
and contents of * that is retrieved with CVS.
I am wondering whether there is a change list or something. If you
tell me here I will update the Wiki ;-)
Thanks
John
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use - on the command line for debugging information, there
should be detailed tracking information provided that will help
On Wed, 5 Jan 2005 12:41:23 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:
> Hi List!
>
> I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to prob
Peter Thanks for your response - have u experimented with the codec
selections, or has anyone?
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Hi,
Anyone used this service, any comments on reliability/support?
Thanks
John
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I'm just about to start implementing this project. I have a test
server working well with SIP phones and IAX for incoming and outgoing,
but when I golive will need 4 analogue lines coming in.
1. Anyone got this config working with a 4 port FXO digium card
2. Any tips/hints/traps
Thanks
John
(My
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