Heya everybody.
I have been peering into the code for zaptel for a while now, I am
keenly interested in the dacs support, being able to apparently
redirect certain spans to other spans. Not sure if this has to be on
the same T1 interface or can be used between T1 interfaces on the same
board or
Just for giggles can you set an absolute timeout in the dialplan for
all calls in and out of that span?
On 1/25/07, kjcsb [EMAIL PROTECTED] wrote:
I have a situation where the two Zap channels on a TDM400 are staying
offhook after a random period of time; it is not (I believe) related to the
I am trying to do a wire level tap on T1 equipment using digum
equipment. So far most call monitoring hardware for call centers try
to stay on the analog side requiring a lot of rewiring. I have
already posted to the list about T1 bridging using DAC's support in
the zaptel drivers. I still
I want to make sure that when an asterisk server dies that I am not
left with a huge bill afterward for not hanging up a long distance
call correctly.
Are digium cards somehow set up to recieve a heartbeat from the
drivers and if it skips a few beats it will take the t1 down in a way
that would
using this solution on many production server whithout problems
It sounds weird but I found it to be very useful with strange zaptel setup
Hope it helps
Regards
Edoardo
Shane Spencer ha scritto:
I want to make sure that when an asterisk server dies that I am not
left with a huge bill
Wow, thanks for the awesome reply :)
On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Shane Spencer wrote:
I am trying to do a wire level tap on T1 equipment using digum
equipment. So far most call monitoring hardware for call centers try
to stay on the analog side requiring a lot
product, as long as I can spy on active
channels somehow. I don't think its going to work that way, I wil
test out libpri for a bit.
Shane+
On 1/28/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Shane Spencer wrote:
I am trying to do a wire level tap on T1 equipment using digum
equipment. So far most
Try setting AbsoluteTimeout() as the first parameter in your dialplan
entry. Check it out on voip-info.org
On 1/28/07, kjcsb [EMAIL PROTECTED] wrote:
Anyway, my question is, how do I get the offhook status to reset? So far
only a server reboot works. I tried:
- physically disconnecting the
I wanted to know if there was a peekaboo factor to it all. You can
flow data under a glass window :)
On 1/29/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
Shane Spencer wrote:
I am very interested in the DACs capabilities of Digium cards, there
is no information anywhere on this. I could always
It was either down or asterisk was frozen. Either way a heartbeat
could fix that.
On 1/29/07, C F [EMAIL PROTECTED] wrote:
Shane, are you trying to say that the PRI was actualy down (the D
channel was NOT up) for the time that ATT is billing you?
On 1/29/07, Shane Spencer [EMAIL PROTECTED
http://www.junghanns.net/en/ISDNguard_produkt.html
I am no longer with the company that got frelled by ATT and/or
asterisk. However this unit would have definitely helped out. Just
disconnect when heartbeat not found.
On 1/29/07, Shane Spencer [EMAIL PROTECTED] wrote:
It was either down
Realtime.. Realtime.. Realtime..
On 1/30/07, j [EMAIL PROTECTED] wrote:
On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote:
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do this, I have to add a couple quick extensions to the
dial plan dynamically, so I
Reload.. Reload.. Reload..
On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote:
Realtime.. Realtime.. Realtime..
On 1/30/07, j [EMAIL PROTECTED] wrote:
On Tue, 2007-01-30 at 23:11 +0200, Tzafrir Cohen wrote:
On Tue, Jan 30, 2007 at 12:59:15PM -0800, chester c young wrote:
In order to do
I'll submit a patch to digium so at least we have
this simple functionality in the future...
j
On Tue, 2007-01-30 at 13:17 -0900, Shane Spencer wrote:
Reload.. Reload.. Reload..
On 1/30/07, Shane Spencer [EMAIL PROTECTED] wrote:
Realtime.. Realtime.. Realtime..
On 1/30/07, j [EMAIL
Cool. My first attempt would have been to find out how to use
asterisk variables in the dialplan since I can set those like crazy
mad via an AGI. Then i would have cried and become horribly
demotivated.
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Reload.. Reload.. Reload..!
/me ducks
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Hahaha,, I think thats a freaking SWEET suggestion :)
On 1/31/07, Andrew Furey [EMAIL PROTECTED] wrote:
On 01/02/07, Yuan LIU [EMAIL PROTECTED] wrote:
What Lee suggested is to have the AGI script to actually parse, insert a new
context in extensions.conf, or deleting from it, then reload
I hate to think of the possible echo if you change the volume and a
sip device didn't know about it. It wouldn't effectively use the echo
cancellation on board, I am not an expert with echo cancellation
however.
It should be possible to just multiply each byte of a waveform by a
percentage if
You need to at least register AFAIK. Download gaim and use its
facilities to rejister. Jabber is not for the faint of heart when it
comes to IM platforms, read up on it if you haven't already.
On 2/3/07, Ian Hailey [EMAIL PROTECTED] wrote:
Hello all,
I am having trouble getting gtalk to work
point to point E1 lines? Or are you interfacing to a PSTN network for
local calling/receiving?
PTP E1
http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration
On 2/5/07, Roger Schreiter [EMAIL PROTECTED] wrote:
Hi,
I'm looking for a mean to send digital data over
an E1 line, just like
I had this happen to me because I was not configured properly and for
some reason the telco was automatically dropping me every so often. I
called the telco and they corrected me.
Shane
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PRI's are sold with SLA's so get them to diagnose your problems using
their own testing equipment. Ideally this would work, however I
typically have to bring a case of beer to my local telco and find my
support team - if that fails I just set it somewhere that makes it
easy for people to trip
I hate to say this, but voip-info.org has a few different methods of
handling this already defined.
If you are 'intercomming' to several styles of SIP based phones, you
have but to only configure the phone to accept those types of calls
and add a SIP header pre Dial().
Shane
do your sip phones dial after a timeout? If the timeout is set to
around 5 seconds you may have a dialplan issue on your sip hardware.
Shane
On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote:
I have had a lot of complaints about the time it takes to setup a call. I
have timed it and it is
I only say this because nobody in our office knew how to use the
checkmark on snom phones to initiate a call, they always just waited
for the phone to initiate the call for them :)
On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote:
do your sip phones dial after a timeout? If the timeout is set
it would be frustrating if the call never terminated
when somebody else answered the queue.
Shane Spencer
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So, watching the asterisk console with full debug on shows something
about Starting Music On Hold for Channel xx/yy-zz?
Shane
On Jan 7, 2008 11:00 AM, Gaëtan Minet [EMAIL PROTECTED] wrote:
Hi
Nobody has an Idea ? Should I try and fill a bug report (or feature request
?) at Digium ?
The
Heya everybody.
I work on a lot of AGI/AMI/AJAM/etc.. projects and recently discovered
RELP (available via rsyslog) which is defined here:
http://www.librelp.com/relp.html
I've been pimping out (yes.. pimping) the Log dialplan application to
quickly emit a message to my local syslog which is
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