On 20/12/05, Douglas Garstang [EMAIL PROTECTED] wrote:
That's probably because I've never received any from you Brian.
Just because it's open source doesn't mean it has to be crap.
Will someone please give this guy a refund so he can go and spend his
money somewhere else.
--
Peter Bowyer
. Considering traditional phone users have come to expect this
functionality, it leaves a lot to be desired as far as Asterisk is concerned.
Off you go to another product then. Close the door on the way out.
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Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP
ever said Asterisk was perfect, and your help with new/missing
features will be welcomed, if you can get over the chip on your
shoulder.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473
'Downloads'
link, tells you exactly what you have to do to obtain the latest
source using SVN.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
: PROPFIND of '/': 200 OK (http://216.27.40.102)
Had to use IP address... no DNS on test box...
Hosts file?
Peter
-
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
as not covered by Csoft.
Shame.Several of the Caribbean networks are covered by Bayham, it
might be worth checking with them if the Anguilla network was missed
off.
Peter
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Peter Bowyer
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).
Peter
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) that I
use their service.
Am I allowed to say that I don't use it anymore?
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to the Asterisk?
I couldn't find a way round this, and ended up using a 'spare' line
presentation on my GXP-2000 phones to register to the voicemail server
simply to pick up the NOTIFYs. Since the phone only has a single MWI
LED, it doesn't matter which line the NOTIFY comes in on.
Peter
--
Peter
want to define this user as type=user, however this can't
work because Asterisk only authenticates users by username, not IP.
Check out 'insecure=very' for sip.conf.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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through - some will give you a static
IP for free.
Time to shop around - most IPStream resellers will migrate you for free, also.
Peter
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Peter Bowyer
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On 08/09/05, Mark Phillips [EMAIL PROTECTED] wrote:
Bloddy 2E's; always wrong.
Mark G7LTT/KC2ENI
I know some G7s who are occasionally wrong, too :-)
Peter G4MJS / 9M6BAA
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Peter Bowyer
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radio is by no means new - there
are many skype-a-like systems around which are used as links or user
access to the existing ham repeater network. I don't know of any using
Asterisk, though.
Peter G4MJS
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Peter Bowyer
Email: [EMAIL PROTECTED]
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that is built in, but ideally this could be accomplished
without using festival because Allison's voice is so much more pleasant.
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SayDigits
Peter
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Peter Bowyer
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.
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to your last posting solve the problem?
Peter
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.
Works fine and dandy here.
Peter
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On 01/07/05, Mark Charlton [EMAIL PROTECTED] wrote:
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
gone away to work on Asterisk
documentation. Seems it's not as important to you as making
incomprehensible postings to mailing lists.
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On 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
Why asterisk.org don't provide a documentation project
?
http://www.asteriskdocs.org/
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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On 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
No i think time spent to work by developpers and users
merit a documentation project .
Please stop typing for a moment and start reading. There is a
documentation project for Asterisk at www.asteriskdocs.org.
Peter
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Email
asking why nobody
has done it yet.
Peter
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. Which is the same answer as you got last time you
asked this question in the past few hours.
Peter
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On 12/06/05, Graham Pearson [EMAIL PROTECTED] wrote:
I am wanting to know where the template file is for the VoiceMail Email
Message. At the present time, the URL Link has a wrong address and I
would like to change this to point to the correct address.
voicemail.conf
--
Peter Bowyer
Email
-- Called 777
Urgent handler
Urgent handler
-- SIP/777-82e9 is ringing
Urgent handler
Any Idea what's wrong --
How is extension 777 defined in extensions.conf? Did you use the stdexten macro?
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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this:
exten = 777,1,macro(stdexten,777,SIP/777)
Peter
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http://lists.digium.com/mailman/listinfo/asterisk
://www.voip-info.org/wiki-Asterisk+Connect+2+servers
Peter
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to launch a URL on the client.
The URL can be sent from (eg) an AGI which can take the callerid info
and do whatever smarts are necessary.
Peter
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Peter Bowyer
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On 27/05/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
The person on 617 is unavailable --- Why
Maybe he's in the bathroom?
The condition being reported is coming from the UA on the end of the
SIP call - is there a DND setting or something there?
Peter
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Peter Bowyer
Email: [EMAIL
a 'pull on demand' model
- no polling or pre-loading of a holding page etc.
Peter
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http
=6000,2,Wait(2)
exten=2000,1,Dial(SIP/${EXTEN})
exten=3000,1,Dial(SIP/${EXTEN})
Did you have a question?
Peter
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to receive alerts for that extension.
And it doesn't run external apps.
Indeed. What would you like it to do? I'm going to play with the
source code sometime soon to address the 'log on' issue, lets' collect
some requirements.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP
want to send the entire command line /
URL from the dialplan?
Peter
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Hi all
What headset do people use with the GXP-2000? Any recommondations for
or against particular models?
Thanks
Peter
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as the wiki page which lists more.
Peter
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. It listens on a TCP port and displays what it
gets sent (with a little special formatting).
It is open source, so I guess it could be hacked to run an external app.
http://sunflowerhead.com/software/yac/
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED
at
asterisk.
can any one tell what is the reason
Did you restart Asterisk - that's a complete restart, not just a 'reload'
Peter
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at an extension matching the individual sipgate
username in the register command.
Works for me and several others
Peter
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:927 dial_exec_full: Unable to
create channel of type 'c' (cause 66)
Have you set the TRUNK variable in the [globals] section of
extensions.conf? Looks like you didn't.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED
to mess around to get the dialstring to end up in the
right format for the provider you're using, also. Or imbed it directly
in the dialplan.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED]
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is simply a convenience in the
dialplan - you don't need to do it that way if you don't want.
Peter
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are optional - custom ringtones and MAC-specific config.
I left my TFTP server pointed to 168.75.215.188, and the phones
upgraded themselves to v 1.0.1.6 without intervention
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED
immediately.
This needs to come from the phone - your phone should have a setting
something like 'unregister on reboot' . Turn this on.
Peter
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Peter Bowyer
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VoIP: sip:[EMAIL PROTECTED]
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these,
they're much better than the BT-100s.
Peter
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in a hash of extensions, the other
sends the callerid information in YAC format.
Email me if you want a copy.
Adam.
p.s. CC adam@mydomain to make sure I see it if you reply.
I'd appreciate a copy of your YAC scripts at your convenience
Regards
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED
to authenticate
with the other server.
The username/secret in iax2.conf is for inbound, not for outbound calls.
Peter
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it with above. See
the difference?
Check the presented username with iax debug enabled to confirm.
Peter
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are on purpose muted by the service
providers or any other reason why it does not work?
I'm not aware of the detailed reason, but DTMF into Asterisk from
Sipgate won't work. This path is well-trodden...
http://www.voipuser.org/forum_topic_844.html amongst other places.
Peter
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Peter Bowyer
Email
- mail vs main - but the bigger problem is that you are
sending calls to a context called 'from-ask-main', but that context
doesn't exist in your extensions.conf. You have one called 'from-sip'
which is where you probably could send them.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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On 22/04/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Fri, 22 Apr 2005, Peter Bowyer wrote:
On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote:
Hello everyone,
I am trying to receive DTMF commands on asterisk from PSTN calls
terminated at my asterisk box. I have tried
to use a local MUA (/sbin/sendmail etc) or talk SMTP directly to an
MTA, either locally or remote. Asterisk voicemail unfortunately is not
one of those systems (AFAICT) - you're stuck with having to use a
local MUA.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL
,
2005 at 01:51:25 AM so you might
voicemail.conf
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the 'third-party' version. http://www.virbiage.com/firefly/download/
Peter
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)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits non-zero.
You need 'Read' instead of 'Background'.
Peter
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VoIP
On Apr 5, 2005 7:45 AM, Matt Riddell [EMAIL PROTECTED] wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial
for.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
Peter
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SIP calls and
direct them to an IVR or a specified extension, for example. But you
probably wouldn't allow them to make toll calls.
Peter
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.
Thanks in advance!
cd /usr/src/asterisk
grep -r voicemail.conf *
should give you a clue or two.
Peter
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server, use
sip show peers
To see what other servers yours has registered with, use
sip show registry
http://www.voip-info.org/wiki-Asterisk+CLI is a useful reference, as
is 'help sip' in the CLI.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED
I believe to be the latest, works A1 on my 15
phones.
Indeed - the bug was fixed in 1.0.5.18 (Nov 14 2004) according to the changelog.
Fixed BT-100 dialing bad URI when using the message button
Peter
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On Wed, 9 Mar 2005 14:13:21 -0700, Wiley Siler [EMAIL PROTECTED] wrote:
Anyone suffering an outage with them right now?
I am getting the following from my box when I try to dial using them.
== No one is available to answer at this time
Look at www.voipjet.com
--
Peter Bowyer
Email
The point is this.
I know where I bought my service and I know where to send email to see
if they say they are online.
I was asking the community to see if anyone else was having a dialing
issue with VoIPJet.
Don't respond if your response is just to be a smartass.
OK.
Peter
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they've fixed it.
Peter
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, 1.0.6 is out...
Peter
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that
those who come later are not left with reading about how you had
exactly the same problem as they're seeing, but they don't know what
you did to fix it...
Peter
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/search?q=asterisk+command+line
leads very quickly to
http://www.voip-info.org/wiki-Asterisk+Starting+and+Stopping
Peter
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it as a compliment that when
it's down occasionally, so many people notice.
Peter
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for - a script to use the
pre-recorded weather terms in the loligo.com extra sounds package :-)
---
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http
On Wed, 16 Feb 2005 03:40:38 +0330, mohammad [EMAIL PROTECTED] wrote:
I saw several examples of Dial app with the format:
Dial(Local/..)
Anybody knows what the Local technology means?
Did you try the WiKi? Or Google?
http://www.google.com/search?q=asterisk+local
--
Peter
pasted this line from your error message into Google:
ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object
and the top result looks to have some good advice for you. Did you try that?
Peter
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you that the remote server is refusing the
connection from your server because of incorrect authentication. Check
the IAX peer/friend entry in the remote server against the credentials
you're using in your friend entry or in the dial string.
Peter
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On Thu, 10 Feb 2005 10:47:22 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
And not to disparage the creator/maintainer of [EMAIL PROTECTED], but you
really need to trust that your install was a little hardened before
placing it on the network.
Indeed. The default root password for a
On Thu, 10 Feb 2005 16:33:46 -0500, Gene Willingham
[EMAIL PROTECTED] wrote:
exten = s,1,Dial(SIP/192.168.1.8:,20); Connect to 192.168.1.8 on
port , with a 20 sec timeout.
exten = s,1,Dial(SIP/[EMAIL PROTECTED]:9876,20,r) ; Connect to sip.com
port 9876, requesting extension
On Sat, 9 Oct 2004 12:56:03 -0400, Steve Totaro
[EMAIL PROTECTED] wrote:
AMP is great and provides integrated extensions to dial to access features
implemented in asterisk. It also provides webbased access to voicemail and
flash panel operator. You will find that many phones have features
On Wed, 9 Feb 2005 18:42:27 -, Mike Wright [EMAIL PROTECTED] wrote:
Unfortunately I seem to have another problem!
I am using sipgate for the incoming line - and it appears that you cannot
get DTMF to work in that configuration. Unless anyone knows anything
different of course!!
I've not
On Thu, 03 Feb 2005 20:02:03 +0100, Stefan Gofferje
[EMAIL PROTECTED] wrote:
Maybe you have something like that too, where your customers don't pay
too much and you don't pay too much. A nice side effect is that nobody
will ever know that your companies HQ is in a lonely little village in
the
On Thu, 13 Jan 2005 10:22:52 +0200, David Norton [EMAIL PROTECTED] wrote:
I am getting this problem when trying to register with Voipfone.co.uk. It
used to work, and I haven't changed anything that I know of.
Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to
On Tue, 4 Jan 2005 16:57:46 +, John Middleton
[EMAIL PROTECTED] wrote:
Anyone used this service, any comments on reliability/support?
Works well for me. A hiccup on initial config was corrected quickly.
Peter
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