Marco wrote:
Hi,
this is my 1st message, I'm writing to ask if anyone knows if a PCI32
card like the TDM400P (quad analog) or the B410P (quad BRI) is working
on a PCI-X bus, at 100MHz or 133 MHz. I'm really stuck with this, since
I found a partial yes on this mailing list but my supplier
Dear Royce;
Did ur problem resolved? Because now I am facing same
problem.
It look like that it happens with IAX trunk only, but
does not happen with IAX endpoints that registering
(as trunk does not register, it sends the call
directly).
My initial analysis that one of the following can help
Tilghman Lesher wrote:
Macros are deprecated. Gosubs are the way forward, and yes, they have
local variables. Simply define them once as Set(LOCAL(foo)=bar) and foo
will be gone when the innermost stack is removed (either by Return or
StackPop).
Yeah, but I like macros.
stamps feet.
Andreas van dem Helge wrote:
I've had the opposite problem. Press mute while the call is still
ringing and it will say MUTE on the display but the microphone is
not muted. It was very embarrassing to discover this bug.
Have you reported this to grandstream? (It's a new one to me, but then
[EMAIL PROTECTED] wrote:
Right now they push the user to buy a 4 channel echo canceller which
you can get from Octasic for $40. The card with 4 ports is retail
around $640.
Err, if you buy a genuine TDM400P (the card we are talking about).
Digium offer a free HPEC license. (although HPEC
On 17/02/2008, Jaap Winius [EMAIL PROTECTED] wrote:
Interesting. I still have several AVM Fritz!Cards, but I stopped using
them after I upgraded my server because I could no longer get them to
work. I used to compile the fcpci module for kernel 2.6.8, but it
doesn't work with 2.6.18. AFAIK,
Because i want a ringing signal while people are in a waiting queue i've
created a wav file containing our local ringing indication
If I make an inside call to the queue, the correct sound is played, but
when i make an external call, no signal is heard.
everything else looks ok, and all other
Hi
I am having problems with Asterisk 1.4.18 and realtime architecture. I use
Postgresql-8.3 as the database.
Everything works OK; all sip phones (their configs are in the database) are
able to register to the server and I can make calls (dialplan is in the
database), but each time Asterisk reads
Can someone tell me how can I monitor asterisk for proxy and redirect
services. The only way I can monitor asterisk now is by using asterisk -r -x
*CLI Command. I need to monitor proxy and redirect service. Can you tell
me how can I achieve it?
Thank you.
On Sunday 17 February 2008 04:44:59 Thomas Kenyon wrote:
That reminds me, I wonder if I can find the receipt for my TDMP400 so
that I can claim my free license.
You don't need your receipt, only the serial number off the sticker on the
card itself.
--
Tilghman
On Sunday 17 February 2008 03:41:09 Thomas Kenyon wrote:
Tilghman Lesher wrote:
Macros are deprecated. Gosubs are the way forward, and yes, they have
local variables. Simply define them once as Set(LOCAL(foo)=bar) and foo
will be gone when the innermost stack is removed (either by Return
On Feb 13, 2008 12:33 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
In the same way that a PHP programmer should not attempt write Python the
way she writes PHP, I would agree with you. However, if you're willing to
adapt to the ways the dialplan works, you can create dialplans which aren't
Quoting Razza [EMAIL PROTECTED]:
I'm running - 2.6.23.15-137.fc8 which appears to be supported at ATrpms.
I built a F8 box, added ATrpms to the repository list, executed yum -y
install fcpci, which forced a kernel upgrade.
Unfortunately, I can't test any of this since I'm running a Debian
Hi All;
If I succeed to receive calls on my FXO ports (Digium)
and still when I place any outsite calls (via the same
ports) I get busy tone coming from the service
provider, what does that mean?
Already the call is routed successfully to the fxo
port that I can receive on it (I can see that on
Hi,
Would anyone have a clue on this issue?
I'm running asterisk 1.4.13. Trying to get a WAV voicemail file
attachment sent to my email address.
Voicemail is working fine. Email notification of a new message works
fine. However, when I set up voicemail.conf to have an attachment of
the
Hi,
On Sun, Feb 17, 2008 at 05:58:50PM +0100, Jaap Winius wrote:
Quoting Razza [EMAIL PROTECTED]:
I installed the drivers/kernel drivers from ATrpms (
http://dl.atrpms.net/all/fcpci-03.11.07-14.fc8.i386.rpm and
... ).
It seems you're in luck to some degree. I downloaded these files and
On Sunday 17 February 2008 10:33:18 John wrote:
On Feb 13, 2008 12:33 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
In the same way that a PHP programmer should not attempt write Python the
way she writes PHP, I would agree with you. However, if you're willing
to adapt to the ways the
Hi all... I did some Google searches and didn't find any info on this so
I'm posting it here... if this was recently discussed, I apologize for the
duplication -- please point me to the appropriate thread.
System Description:
Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard
Quoting Axel Thimm [EMAIL PROTECTED]:
There are patches inside that will work on Debian as well, just get
the src.rpm and pick out the patches.
Now, why am I not surprised? Actually, if I had known this back in
early December, I'd be following your advice and thanking you now. I
previously
Hello all,
I am struggling with sending voicemail as an attachement in Email.
When i have given the email like [EMAIL PROTECTED] it is delivering
to my gamil account perfectly(of course to spam folder).
But when i given the email like [EMAIL PROTECTED] it is not
delivering to my company email
I am trying to send 'codes' over an isdn2 link - such as *#24# - to
activate call forwarding.
But it doesn't work. I have tried sending it as a straight dial, and
also as a DTMF string...but no luck...
I spoke to a telco tech and he said I had to send a facility
codehuh?
Anyone
hi.
could somebody explain how exactly the following parameters
in zapata.conf work:
pridialplan
prilocaldialplan
internationalprefix
nationalprefix
localprefix
privateprefix
unknownprefix
the wiki comments doesn't quite explain them. and
phone companies are absolutely no help.
i've setup
Hi all,
I have configured my asterisk server as gateway and gatekeeper both.
I am trying to call using SIP agent to h.323 agent but it is not successful.
I have configured ooh323.conf as
gateway=yes
gatekeeper=10.17.112.12
Still its not working.
What configuration file I need to change
Problem:
When I have more than one IAX2 connection (on server zuiderven), I have
problems in receiving calls from IAX peers except for the first in the
list as seen by the iax2 show peers command.
In my tests it showed that by removing one by one the entries from the
iax.conf file in the order as
Hello
I'm using mfcr2 support (unicall) in asterisk 1.4. Everything is working fine,
asterisk can answer calls.
But after some random period of time mfcr2 module stuck. When I make a call to
my * box I can hear only signal of getting caller ID (tritirirti - like
jumping on :) ) and connection
On Mon, 18 Feb 2008, Paul Hales wrote:
I am trying to send 'codes' over an isdn2 link - such as *#24# - to
activate call forwarding.
But it doesn't work. I have tried sending it as a straight dial, and
also as a DTMF string...but no luck...
I spoke to a telco tech and he said I had to send
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