Hi friends,
I have one customers as originator but he have more than 3 IPs, I need to
assign his all IPs under one account type peer.?
i tried the following but it did not work
[vps]
type=peer
context=from-vps
host=209.85.24.98, 67.19.85.130, sip.qualityfone.com
allow=all
Could you please
Vincent wrote:
Hello
I've never built an IVR before, so I was wondering if someone
could share some code from their extensions.conf that would perform
some of thoses steps:
snip /
Hi Vincent.
Look back a few hours in this mailing list for the message called
IAX2: Incoming calls
Hi
I have asterisk ip-pbx on my network, with some grandstream ip phone
and i have cisco gateway that is connetced to VOIP service providers .
Cisco is 3700 series and is using H323 .
i have compiled H323 on asterisk . now i want to make a call from ip
phone that is registerd to
On Fri, 2007-10-19 at 18:04 -0400, [EMAIL PROTECTED] wrote:
SNIP
Can you use .call files I have an approx 1kb PHP script that can be
used for click to call
SNIP
Why use .call files when Asterisk Manager will allow direct interfacing?
I might be interested in doing this, but would prefer
Philipp Kempgen wrote:
http://www.asterisk.org/node/48325
http://www.asterisk.org/node/48360
Brilliant, that works a treat, thanks! :)
Now, for my next question
I have 2 remote sites; 1 @ home, and 1 which I will shortly be transporting
to Spain. I've already set up my dialplan so
Anthony Rodgers wrote:
We tried with MS Exchange but couldn't get it to work (MS Exchange
doesn't support a master account).
It used to? Not out the box though...
Ed W
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
I wrote:
So, is there any way of monitoring the status of a device on
a remote
server, perhaps utilising an IAX2 channel?
To answer my own question, I did it :)
In case anyone's remotely interested in a similar setup/idea, here's the
relevant bits of my dialplan.
Assume that a call,
Michael J. Liberatore wrote:
Well this is the bug I am having with the make install of 1.4.5.1:
http://bugs.digium.com/view.php?id=10156
Even though I got it to install ztcfg -vvv still says 1.4.4 also.
Mike
We just made a new zaptel release (1.4.6) in which there were many
fixes.
[EMAIL PROTECTED] wrote:
The SPA921 config has a NAT Keep Alive Intvl which is set to 15 by
default, which I'm taking to mean it has NAT keep alives enabled.
No, look under the Line 1 or Line 2 tab
Found it - thanks again.
Whilst I've got your attention - have you managed to make an
Per Jessen wrote:
Whilst I've got your attention - have you managed to make an SPA do a
periodic config refresh? As far as I can tell, mine is all set to go,
except it doesn't. I can poke it with the admin/resync URL, but I'd
rather that it would ask on its own.
I forgot to add - using
I was previous using Asterisk 1.2.9.1 and decided to get some real servers
outside of my house. It was time for Asterisk 1.4.4.
I figured since all the conf files were in /etc/asterisk form the old box,
i'd just copy tha directory over to the new server. My SIP DID AGI stuff
worked, except
I hope no one will frown on my post here as our product is commercial.
I just wanted to let you know that you can use Thirdlane PBX Manager to
create complex dialplans. The way it works is that you can create scripts -
equivalent of Asterisk macros with some extras and no limitation on what
On 10/20/07, Per Jessen [EMAIL PROTECTED] wrote:
Found it - thanks again.
Whilst I've got your attention - have you managed to make an SPA do a
periodic config refresh? As far as I can tell, mine is all set to go,
except it doesn't. I can poke it with the admin/resync URL, but I'd
rather
On 10/19/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I know I've said this time and time again, but just for the purpose that
this will be archived somewhere on the net, there should not be any more
problems related to interrupts and specific servers. If there are,
*please* let me know
On 10/19/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
Are you using Asterisk 1.4 ?
If positive, are you then successfully using IMAP storage ?
Your input would be very valuable to decide if rewite of IMAP storage could
be considered as bug fix (non one uses IMAP now) or as a new feature
On 10/20/07, Dave Walker [EMAIL PROTECTED] wrote:
On Fri, 2007-10-19 at 18:04 -0400, [EMAIL PROTECTED] wrote:
SNIP
Can you use .call files I have an approx 1kb PHP script that can be
used for click to call
SNIP
Why use .call files when Asterisk Manager will allow direct interfacing?
On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
--
Erik Anderson
http://andersonfam.org
[EMAIL PROTECTED] wrote:
On 10/19/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I know I've said this time and time again, but just for the purpose that
this will be archived somewhere on the net, there should not be any more
problems related to interrupts and specific servers. If there
Hello,
Not sure if this is what you are looking for but we've played around with
JIAXClient and wrapping it with Javascript and Flex/Actionscript to create a
custom UI (e.g. button, flash).
You can take a look at a demo here (
http://code.google.com/p/blindside/wiki/Click2Call)
Richard
On
Alex,
In my sip.conf, I have some lines that i need to register on my voip
provider and I have this configuration:
register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
How can I hide or use md5 for this configuration ?
Thanks.
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
On 10/20/07, Abdul [EMAIL PROTECTED] wrote:
Hi friends,
I have one customers as originator but he have more than 3 IPs, I need to
assign his all IPs under one account type peer.?
i tried the following but it did not work
[vps]
type=peer
context=from-vps
host=209.85.24.98, 67.19.85.130,
On 10/19/07, Alan Lord [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
So your problem is:
-- IAX2/alanb-3 answered SIP/101-081d1050
Except the remote end didn't actually answer the call? The problem is
your remote end... its answering the call. All the IAX hardphones I've
this message is basically tells you asterisk is not running.
can you check and see if asterisk is running and present in memory?
something like
ps -ef | grep asterisk
On 10/20/07, Dominic Son [EMAIL PROTECTED] wrote:
I was previous using Asterisk 1.2.9.1 and decided to get some real
servers
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .
On 20/10/2007, Al lists [EMAIL PROTECTED] wrote:
this message is basically tells you asterisk is not running.
can you check and see
1.4.6 fixed the make install problem but has broken my zaptel
completely. When starting I get:
Loading zaptel hardware modules: wctdmNo functioning zap hardware found
in /proc/zaptel, loading ztdummy
Running ztcfg: .
/proc/zaptel is empty.
Running ztcfg -vvv gives me:
Zaptel Version: 1.4.4
On Sat, Oct 20, 2007 at 09:46:57AM -0500, Matthew Fredrickson wrote:
Michael J. Liberatore wrote:
Well this is the bug I am having with the make install of 1.4.5.1:
http://bugs.digium.com/view.php?id=10156
Even though I got it to install ztcfg -vvv still says 1.4.4 also.
Mike
On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote:
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .
chmod 777 (or even 666) to the control socket (asterisk.ctl)
On Sat, Oct 20, 2007 at 02:52:06PM -0400, Michael J. Liberatore wrote:
1.4.6 fixed the make install problem but has broken my zaptel
completely. When starting I get:
Loading zaptel hardware modules: wctdmNo functioning zap hardware found
in /proc/zaptel, loading ztdummy
Running ztcfg: .
awesome. it worked. thanks guys.
On 10/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Oct 20, 2007 at 11:57:05PM +0530, Jaswinder Singh wrote:
astrundir = /var/run
Change this to astrundir = /var/run/asterisk on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create
I did get the old version running again, you were correct, it was still
loaded, I realized that right after I sent the first message.
I am running debian sarge with 2.6.15.4 with devfs. Not 2.6.8 though.
Could this be a problem with my system and asterisk/zaptel in general?
I have had countless
On Sat, Oct 20, 2007 at 10:17:02PM -0400, Michael J. Liberatore wrote:
I did get the old version running again, you were correct, it was still
loaded, I realized that right after I sent the first message.
I am running debian sarge with 2.6.15.4 with devfs. Not 2.6.8 though.
Could this be a
Note: forwarded message attached.
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Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com ---BeginMessage---
Dear Guys,
Recently, I have done an update on my DG-104S with Boot Prom
Isee no one has answered you so far, so I'll try:
I have asterisk ip-pbx on my network, with some grandstream ip phone
and i have cisco gateway that is connetced to VOIP service providers .
Cisco is 3700 series and is using H323 .
i have compiled H323 on asterisk . now i want to
Erik Anderson wrote:
On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
If you are trying to use non-complied (XML) profiles... don't even
bother wasting your time.
Why is that? I'm using the xml-style config and they're working just fine.
I'd like to be able to
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