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
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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for.
http://www.voip-info.org/wiki-Asterisk+cmd+DISA
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
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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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
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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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 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 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, 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
for - a script to use the
pre-recorded weather terms in the loligo.com extra sounds package :-)
---
Peter Bowyer
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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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Tel
it as a compliment that when
it's down occasionally, so many people notice.
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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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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, 1.0.6 is out...
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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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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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
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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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by registering as another user but
Asterisk doesn't allow multiple registrations.
Which part of Asterisk?
register = nn:[EMAIL PROTECTED]/m
register = oo:[EMAIL PROTECTED]/
Works fine for me
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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to the
Exchange server with authentication.
http://msmtp.sourceforge.net/
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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)
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: [EMAIL PROTECTED]
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?
something.pls isn't a stream, it's a playlist which (probably) lists
streams within it.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
(SIP/2003)
Give me a shout if you want more help
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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it
exists, how can you all benefit from this ?
asterisk-biz is the correct place. This isn't.
Peter
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aggregators that I could use for sending messages to this
particular phone over the Internet?
There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk.
bayhamsystems.com have a service tailored for Asterisk users.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED
this all the time, and never bothers to respond to objections.
Doesn't answer questions about how he mis-describes his products,
either.
Peter
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On 14/02/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Peter Bowyer wrote:
There are plenty - I've used 2sms.com, clickatell.com and csoft.co.uk.
bayhamsystems.com have a service tailored for Asterisk users.
These are all based in the UK. What if I'm in North America?
Does it matter?
What
?
Have you experienced any difficulty asking or answering questions
about Asterisk 1.4 here?
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account
for each and an entry in the dialplan which rings both.
Peter
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seconds? A minute if
you're a slow typist...
Yes, you can do this. #include is a literal text include, as the last
poster said.
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on.
All of which, to repeat, could be experienced with a small investment
of your time. It really does pay to experiment with the simple things,
you find your learning curve is so much flatter than if you ask
questions in a vacuum.
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED
all that for free. Enjoy!
Peter
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On 06/10/06, ram [EMAIL PROTECTED] wrote:
Hi
can some one clarify
does the aterisks act like a SER
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy
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perhaps the rest of your dialplan is expecting the call to come in
with a destination which matches your DID - in which case, put the DID
number there instead of the 101.
Peter
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On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.
And that new-fangled electricity will never catch on - lets stick with
gas-lamps...
--
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Email: [EMAIL PROTECTED
On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote:
On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.
And that new-fangled electricity
a sip debug and see what it's telling you about the call, post it
here if it doesn't help.
Peter
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of staying
with a supportable version of software, especially if it's open
source. If there's a security-related bug found in your version, will
it get patched, or will you have a forced upgrade several versions
ahead on your hands in a hurry?
Peter
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Peter Bowyer
Email: [EMAIL PROTECTED
to your last posting solve the problem?
Peter
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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
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL PROTECTED
,
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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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
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Peter Bowyer
Email: [EMAIL PROTECTED]
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VoIP: sip:[EMAIL
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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- 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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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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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
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
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these,
they're much better than the BT-100s.
Peter
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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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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
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
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Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
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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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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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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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]
Tel: +44 1296 768003
VoIP
want to send the entire command line /
URL from the dialplan?
Peter
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=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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a 'pull on demand' model
- no polling or pre-loading of a holding page etc.
Peter
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http
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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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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Email: [EMAIL
://www.voip-info.org/wiki-Asterisk+Connect+2+servers
Peter
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-- 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
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VoIP: sip:[EMAIL
this:
exten = 777,1,macro(stdexten,777,SIP/777)
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
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Peter Bowyer
Email
. 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 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
Why asterisk.org don't provide a documentation project
?
http://www.asteriskdocs.org/
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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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asking why nobody
has done it yet.
Peter
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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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.
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
switches or some other technique),
then calls to that extension will be handled by the server it registered
to automatically.
Use an IAX2 switch for a small, known number of servers. Consider
DUNDi to extend into a larger, more dynamic 'cloud'.
Peter
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Peter Bowyer
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be able to
happen.
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FWD: **275*5048707000
VoipTalk: **473*5048707000
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position to
take - your call - declaring it as useless without giving it a try,
however, is not really helpful or accurate).
Others are busy finding it very useful indeed.
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FWD: **275
to you a
few days ago?
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you think it
won't work. Then we can help you get it going.
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Peter Bowyer
Email: [EMAIL PROTECTED]
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1000 phones re-registering
every minute would be like? That's SEVENTEEN new registrations per second.
Yeah - sounds a lot, but only when you're watching a SIP debug. What
traffic do 1000 PCs produce against a Windows server? I wouldn't
discount it simply on that factor.
Peter
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Peter Bowyer
package onto their
Enterprise Server, they say na!
Then shouldn't you be requesting support from the supplier of that
audited, stable platform, instead of requesting community support?
Isn't that why you (they) bought it?
No much point otherwise.
Peter
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