We are running the default asterisk package on Ubuntu Dapper. Our
connection to the PSTN is over an IAX trunk with our provider. We are
getting really bad call quality on calls over the IAX trunk--voice
seems to be garbled or out of order and often completely breaks up.
But on internal calls
We are running the default asterisk package on Ubuntu Dapper (which
has the advanced timing options used by ztdummy). Our connection to
the PSTN is over an IAX trunk with our provider. We are getting
really bad call quality on calls over the IAX trunk--voice seems to be
garbled or out of order
I'm running an asterisk server that uses an IAX2 trunk with our voip
provider for its PSTN gateway. We have no telephony hardware in our
server. We are consistently getting 99.975% or better when we run
zttest. I have heard that this is bad in some cases, but I'm
wondering if it matters, since
of the dialplan.
On 6/16/06, Doug Lytle [EMAIL PROTECTED] wrote:
Carl Youngblood wrote:
No, ${EXTEN} contains i at that point in the dialplan.
exten = 123,1,Set(_TMPEXTEN=${EXTEN})
exten = i,1,SayDigit({$TEMPEXTEN})
You need to read the document in the Asterisk source directory on the
subject of variable
No, ${EXTEN} contains i at that point in the dialplan.
On 16 Jun 2006 01:41:30 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Does SayDigits(${EXTEN}) not work in this case? I would imagine that it would
still maintain the dialled extension in that variable, would it not?
Undrhil
---
there.
Thanks,
Steve Totaro
Carl Youngblood wrote:
Of course I'm trying to deal with the network problems, but it's nice
to have another method of verifying that everything is working.
Frequently there are people who don't complain, so we don't realize
that their call quality is sub-par.
We
Thank you! Thank you! I had been trying all sorts of convoluted ways
to get that information. That was very easy.
On 6/16/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Variables
Use ${INVALID_EXTEN}
On 6/15/06, Carl Youngblood
I am trying to modify a fairly complex digital receptionist dialplan
that has a number of included contexts. Right now the system is not
announcing the extension that the caller attempted to dial, so callers
get confused when they think they dialed a valid extension but
asterisk didn't pick
I have been getting occasional reports of dropped calls from the users
of our asterisk system. Is there anything I can monitor in my logs or
in the console to see when a call is dropped? I'd like to see if
these drops coincide with network traffic problems.
Thanks,
Carl
somewhere if a call was
terminated abnormally.
Thanks,
Carl
On 6/14/06, Steve Totaro [EMAIL PROTECTED] wrote:
Carl Youngblood wrote:
I have been getting occasional reports of dropped calls from the users
of our asterisk system. Is there anything I can monitor in my logs or
in the console
Our asterisk system gains access to the PSTN through a voip provider.
We have no digium or other telephony hardware in our system. Do the
zttest results still matter to us? Our results were as follows:
--- Results after 1007 passes ---
Best: 100.00 -- Worst: 99.780273 -- Average: 99.975763
Thanks. What is it in the 2.6.13-based kernel that improves timing?
Should I expect to see a significant improvement if I upgrade to it?
On 6/13/06, Mike Fedyk [EMAIL PROTECTED] wrote:
IAX trunking and meetme conferences are some of the heaviest users of
zaptel timing. I'd suggest if you
FYI, the digit timeout was simply too short in my IVR. After
increasing that everything worked fine. This problem only showed up
on cell phones, many of which don't allow you to type long digits so
your keypresses have more silence in between them.
On 5/12/06, Carl Youngblood [EMAIL PROTECTED
I'm having a big problem where digits dialed from certain cell phones
are too short to be recognized by my asterisk server. I'm running AAH
2.8. Some cell phones don't allow the caller to hold down the digits
and have the tones play as long as they hold them down for. They just
play a short
Thanks to everyone who responded. I was able to modify the freepbx
paging code to use something like the suggested macro and it worked
well. For those who may be interested, the following Page macro works
for Linksys SPA942 phones:
[macro-page];
;
; Paging macro:
;
; Check to see if SIP device
I have gotten intercom working on my office phones (Linksys SPA-942s),
but I have noticed that if someone is in a call, it places the call on
hold and sends the intercom audio to the person holding the phone that
is being paged. I'd like to add logic to my dialplan that doesn't
send a page to a
I am running AAH 2.8. I have an IVR for our main phone number that
allows users to dial an extension directly. I would like to have a
this call may be recorded announcement played before the call gets
transferred. There is not a built-in option for this in the IVR web
interface, but one way I
Hi Jeff,
I live in Provo and I think I understand the application you're
referring to. Some folks in my neighborhood have been getting to be the
beta testers for these cool new fiber links that the city is supposed to
be laying out. If I only lived a few blocks over, I would be able to
get
hardware. Does a service like this let
you make more than one phone call simultaneously, or must you pay an
additional $8.00 for each line that gets used at the same time? Sorry
if these questions are stupid but I'm new to asterisk.
Thanks,
Carl Youngblood
What about the G.729 codec? From what I've heard it allows you to
stuff an analog call into 8 Kbps. This would give you a theoretical
maximum of 80 simultaneous connections on a 640 Kbps DSL line. I
would expect this to be much lower in practice, say 20 simultaneous
streams, but still,
From what I read here:
http://www.globalipsound.com/pdf/gips_iLBC.pdf
iLBC is free and better quality than G.729A, same quality as G.729E and
offers substantially better quality over congested networks. Its
bandwidth requirements are a little higher (13-15 kbps) but they aren't bad.
Adam Hart
Sorry, but would someone mind giving a brief explanation to newbies as
to why this is cool? I am interested in creating call trees from a
postgres database, so this looks like it might be useful, but I still
don't understand much of what's going on here.
Thanks,
Carl Youngblood
On Dec 6, 2003
.
www.deltathree.com
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 05, 2003 1:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internet-to-phone gateway?
On Thu, 4 Dec 2003, Carl
a server with an internet connection and someone else
provides the internet-to-phone gateway. Come to think of it, would
asterisk even be needed for this kind of a solution? Does anybody do
this?
Thanks,
Carl Youngblood
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Anyway, if anyone could point out any work that has already been done in
this regard, I would really appreciate it.
Thanks,
Carl Youngblood
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[EMAIL PROTECTED]
http
Just a warning that you will not know when the line is picked up with a
X100P. Later when you upgrade to T1/E1 service you will know when it is
picked up.
So I assume that means that I should just wait until I hear some level
of noise and then I know that the line has been picked up?
You may
What is EAGI? I will probably use festival for the time being, but I
thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the likelike TTS, that sucks up
Or maybe noise would have to last for more than a certain period of time
before it triggered another waiting sequence. Like, say, if noise lasts
for longer than 2 full seconds or something.
That may be fine. Although you may have trouble with some line that also
is feeding back echo. That
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