sorry for my delayed answer. I was guessing it was audio conversion becuase it
only happens when the call involves PSTN and that is the most common problem.
My next guess would probably be interupts, but I saw someone already replied
with that. I would also check how you have echo cancellation
Hi,
I tried thousands of time and finally I am a step closer to the solution.
I recompile iksemel with the option --prefix=/usr
I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4, re-extracting
everything from the tar
recompile everything and now jabber is working or almost. When I
On Fri, 9 Feb 2007, Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Question is how does asterisk detect the call without answering?
I'm not wanting Asterisk to handle the call if it is a fax if
i have a full fxo TDM24 and i have problem with installing it.
when i run modprobe wctdm24xxp dmesg shows the following messages.
and it waits for ever and nothing will happen.
i'm sure that:
- the power is plugged into tdm24 board
- udev is configured and is working with other tdm cards.
-
Hi,
As far as I understand,
Fax detection works by the fax machine sending a specific tone down the
line.
Until the phone has been answered, there is no audio path so the receiving
end cannot know what type of call it is.
The stand-alone fax switch devices that appear from the calling end to
Rilawich Ango wrote:
Noah,
Thanks for you reply. I have a problem in call parking as following.
scenario 1
1.Caller A - callee B
2.Callee B answered
3.callee B dial # to park the call and hear transfer
4.callee B dial 700 to park the call
5.callee B hang up and caller A hear 701
Why
Hi,
I am new to Asterisk and am runing asterisk 1.2.9.1 on an OpenBSD box. With a
few manuals I was able to set up some SIP providers with which outgoing and
incoming calls work. However, there is one provider with which inbound calls
don't work at all.
The only apparent error/warning message is
Lee Howard wrote:
Yes, I do suspect that Digium sees things this way.
Maybe I'm too much of a free-thinker - too believing in the open-source
philosophy, but I would like to think that this is not neccesarily
true. I would like to think that they could host and support a
non-disclaimed
Hi Gordon
Following you dial plan
How does Asterisk know to move from s,2, to either incoming,1, or fax,1,
The only jump I recognize it Goto(internal,incoming,1) which should take
all calls to incoming,1, and not fax,1,
OT: is spandsp rxfax handled by astlinux ?
Thanks again
Barry
Gordon
On Sat, Feb 10, 2007 at 06:12:03AM -0600, Lacy Moore wrote:
Lee Howard wrote:
Certainly I think that it's fair to say that some contributions will not
be disclaimed in the scenario I outlined that would have been disclaimed
in the present scenario. I think that depends on how well Digium
Am 10.02.2007 um 14:06 schrieb Tzafrir Cohen:
No. RedHat publish the full sources (as easily-rebuildable source
packages) to all the packages in RHEL. This is why CentOS is possible.
Digium may just as well *bundle* code of that sort in Asterisk
(e.g: as
a separate AGI script, or whatever).
[EMAIL PROTECTED] wrote:
I am looking for some Linksys and GrandStream ATAs in Canada. I am
looking for places that ship from Canada so I don't have to deal
with the clearing of customs and tax remittance.
Any suggestion?
We have all Linksys and Grandstream phones and ATAs in stock and
Benito:
To someone who have done the dCAP exam.
I would like to know about it: test and practises questions examples,
difficulty level,... I'll be very grateful if somebody sends me an
exam model.
When are you planning to write it?
I have just written it (yesterday) but do not yet have my
Justin Newman wrote:
We have considered working on this. T38 is a short term solution, though.
Justin Newman
Why would it be interesting to you to implement T.38? It seems you are
also someone who doesn't disclaim code and get it into SVN.
Steve
--
From:
On Sat, Feb 10, 2007 at 02:57:51PM +0100, Stefan Wintermeyer wrote:
Am 10.02.2007 um 14:06 schrieb Tzafrir Cohen:
No. RedHat publish the full sources (as easily-rebuildable source
packages) to all the packages in RHEL. This is why CentOS is possible.
Digium may just as well *bundle* code of
Hi,
I've been working on migrating my asterisk from zap to sip (due to
compatibility issues between my TDM400P and my Hauppauge PVR500). I've
purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP
phone). I managed to get it all working with my asterisk 1.4.0 installation,
BA == Benny Amorsen [EMAIL PROTECTED] writes:
BA I reported this bug in much more detail in bugs.digium.com, but
BA the bug is gone now without even an email saying where it went. I
BA don't remember the issue number. Somewhat frustrating.
Yay my bug is there after all! ID is 9020. It doesn't
I'm using the Mandriva (cooker/contrib) asterisk-plugins-jabber package.
With this, asterisk is able to sign in to my gmail account. However, I'm not
able (from an other gmail account, using google talk) able to setup a
phonecall. The gtalk / jabber modules have crashed asterisk a few times.
I'd
Are there 45 G.729 instances for the 45 ZAP legs in addition to 45
G.729 instances for the 45 SIP legs? Or do the ZAP legs not get a codec
(HW instead)?
On Sat, 2007-02-10 at 12:06 -0500, Andres wrote:
Hi Matthew,
Yes, those are really 90 SIP-ZAP calls. Which means the 4 port T1 is
Hi folks.. just a few weeks ago I wrote this to someone else:
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put ETH0 on a seperate IRQ from the
What is the difference between using my proprietary asterisk-add on
than to using my proprietary email client (Microsoft Outlook) with my
GPL IMAP servers? You guys need to drop your BS elitist point of view,
It isn't your software, its talking to your software like any other
software does, the
Stefan,
When I have 2 SIP endpoints that both aren't configured with
canreinvite=no then I get no sound.
The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
especially when there is no NAT involved.
On Sat, 10 Feb 2007, Matt wrote:
Hi folks.. just a few weeks ago I wrote this to someone else:
We have several 2900s in production as VoIP servers.. no lockups.
On every server I go into the BIOS and:
* Disable USB
* Disabled uneeded things like Parallel, Serial
* Put
For my next servers I'll be ordering Arima mainboards I think and assemble
the things myself again.
I'm thinking about going with MBX.The problem with this is we were
replacing an IBM server that had the same problem. The Dell Techs all
assured me this is how all the new machines
On Sat, 2007-02-10 at 03:14 -0500, Il Neofita wrote:
Hi,
I tried thousands of time and finally I am a step closer to the
solution.
I recompile iksemel with the option --prefix=/usr
I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4,
re-extracting everything from the tar
recompile
I'm not here to flame anybody. Please see the replies in-line.
Try to actually read them.
On Sat, Feb 10, 2007 at 01:19:30PM -0500, Andrew Joakimsen wrote:
On 2/10/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
No. RedHat publish the full sources (as easily-rebuildable source
packages) to all
On 2/10/07, Luki [EMAIL PROTECTED] wrote:
Stefan,
When I have 2 SIP endpoints that both aren't configured with
canreinvite=no then I get no sound.
The Sipura 3102 definitely works fine with canreinvite=yes and I never
really had a problem with any of the Sipura devices in this respect,
Matt wrote:
For my next servers I'll be ordering Arima mainboards I think and
assemble
the things myself again.
I'm thinking about going with MBX.The problem with this is we were
replacing an IBM server that had the same problem. The Dell Techs all
assured me this is
On 2/10/07, Patrick [EMAIL PROTECTED] wrote:
Where can I find that patch?
Thanks,
Patrick
Hi Patrick,
I downloaded the patch from here
http://bugs.digium.com/view.php?id=7764
___
--Bandwidth and Colocation provided by Easynews.com --
try booting with APIC and ACPI disabled?
Thats right. I have never seen a shared IRQ with Dell servers using
APIC. A RHL ES3 by default enables APIC so I have never even had to
fiddle around with it.
Andres.
___
--Bandwidth and Colocation
Matthew Rubenstein wrote:
Are there 45 G.729 instances for the 45 ZAP legs in addition to 45
G.729 instances for the 45 SIP legs? Or do the ZAP legs not get a codec
(HW instead)?
Its a 4 Port T1 with 92 ZAP channels. So we are talking about 90 SIP
Channels being fed into 90 ZAP
Yes, that is perfectly clear - thanks for the data. The problem with
discussing load capacity of hosts running codecs is just how many codecs
are running at a time, how many code/decode instances each call
comprises, without knowing how many codecs are running per call.
I'm
On Sat, 10 Feb 2007, Andres wrote:
try booting with APIC and ACPI disabled?
Thats right. I have never seen a shared IRQ with Dell servers using APIC. A
RHL ES3 by default enables APIC so I have never even had to fiddle around
with it.
Ofcourse you don't. But simply because APIC makes
try booting with APIC and ACPI disabled?
ARG. You're going around in circles as bad as the Dell tech. If the IRQs
are being shared and setup in BIOS.. no amount of booting with ACPI
disabled, or trying to set the IRQ within Linux is going to help.At this
point, the BIOS is overriding
Thats right. I have never seen a shared IRQ with Dell servers using
APIC. A RHL ES3 by default enables APIC so I have never even had to
fiddle around with it.
Possibly because you are using virtual IRQs... hence none of them appear to
be shared in Linux. Check the BIOS to see what's really
Andres wrote:
try booting with APIC and ACPI disabled?
Thats right. I have never seen a shared IRQ with Dell servers using
APIC. A RHL ES3 by default enables APIC so I have never even had to
fiddle around with it.
Andres.
___
--Bandwidth and
Matt wrote:
try booting with APIC and ACPI disabled?
ARG. You're going around in circles as bad as the Dell tech. If the
IRQs are being shared and setup in BIOS.. no amount of booting with ACPI
disabled, or trying to set the IRQ within Linux is going to help.At
this point, the
We'll see what happens on Monday. I e-mailed several people in Dell who
were involved in this sale and all assured me it would be fine and work.
Needless to say, I am not happy at all right now.
On 2/10/07, Andrew D Kirch [EMAIL PROTECTED] wrote:
Matt wrote:
try booting with APIC and
On Sat, 10 Feb 2007, Barry Fawthrop wrote:
Hi Gordon
Following you dial plan
How does Asterisk know to move from s,2, to either incoming,1, or fax,1,
The only jump I recognize it Goto(internal,incoming,1) which should take all
calls to incoming,1, and not fax,1,
In this particular
Correct, if you want do do *8 + exten you need to use the dialplan and
the Pickup() applicaiton.
On 2/9/07, John Breen [EMAIL PROTECTED] wrote:
Ken Williams wrote:
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don't work's (*8EXTEN)
and when
Well, I've been e-mailing back and forth with a Dell techrep. (This is in
addition to the e-mail I sent to the sales guys, who are not in today).
The techrep basically said:
Your TDM2400P card is only PCI and not PCIe; and, is therfore not
compatible.
According to him, and he backed everything
M == Matt [EMAIL PROTECTED] writes:
M I talked to Dell technical support and they said oh all our new
M machines share IRQs like that, the way you are trying to do it is
M archaic.
The technical support is right. Digium should fix their driver (or
possibly the card). Perhaps it's fixed in the
M == Matt [EMAIL PROTECTED] writes:
M According to him, and he backed everything up (and I have no doubts
M about how PCIe is working) with PCIe devices can share a single
M IRQ, because there is so much more throughput. However, obviously
M with PCI this does not work.
Why does it not work
Hello
Before I order a Travla C156 case
(http://206.14.132.88/products/Travla/c156/C156.html), a Via mini-ITX
motherboard (either the fanless ME6000
http://idotpc.com/TheStore/pc/viewPrd.asp?idcategory=50idproduct=4 or the
fan-equipped M1
I have built several mini-itx via systems, and they work fine with
Asterisk.
We use one with a dual port E1 card in our office as our phone system.
PaulH
On Sun, 2007-02-11 at 00:47 +0100, Vincent Delporte wrote:
Hello
Before I order a Travla C156 case
Why does it not work with PCI? Digium cards are doing 1000 interrupts
a second. That's basically the same as no interrupts at all,
load-wise. (It's also pointless; it would be better to pick a
different timer and turn interrupts off entirely. Admittedly Linux
hasn't been ready for that kind of
Matt wrote:
I guess the question is... is it even possible to have a real-time
VoIP card running on PCIe? Or with 1,000 Interrupts a second.. does
it simply need to have its own IRQ?
Have you tried the Sangoma PCIe cards?
APIC is supposed to fixed the PCI IRQ problem. AFAIK, APIC is not a
Hugh L. Johnson wrote:
I have a Dish 301 receiver that will not display CallerID when connected
to FXS module on TDM400. Uniden phone connected to the same FXS module
does display CallerID.
When Dish 301 receiver is connected to IAXy CallerID is displayed
properly.
Any suggestions on getting
So you want the caller ID for the user that is recieving the call to be the
caller ID for the extension that was dialed for them. Is this correct ? If
so then you can just change the caller ID in the dial plan with Set
CallerID. Have a look here:
hi
I am using asterisk in my production environment and i have a problem
recording the calls.
I use Monitor command to record the calls in to the same server as asterisk
is running.
Some recorded calls have a problem , like the in channel voice recorded in
delay so the entire recorded
to do it is archaic. What?!?! The Dell tech guy kept saying that
I can
define an IRQ in Linux, and I kept telling him that I need two unique
(not
Doesn't IO-APIC work for you or is that what you meant by virtual IRQ?
I thought IO-APIC changed the way the APIC worked but it was under OS
hmmm - we have a recording system running that records up to 60 calls
per second with no issues.
We have a 4 port digium E1 card in each box - what hardware are you
using?
PaulH
On Sat, 2007-02-10 at 21:58 -0500, [EMAIL PROTECTED] wrote:
hi
I am using asterisk in my production
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part I can't seem to get right is the dial function. I tried to
mimic the dial plan
http://www.canadianvoipstore.com/home.php
VOIPSupply has a candian store-front. I bought from VoipSupply and
products shipped from Canada...
Todd
On Feb 8, 2007, at 1:47 PM, [EMAIL PROTECTED] wrote:
I am looking for some Linksys and GrandStream ATAs in Canada. I am
looking for places
From: Roy Kidder [EMAIL PROTECTED]
Date: Sat, 10 Feb 2007 23:15:07 -0500 (EST)
I'm writing an AGI script and want it to dial a number on a channel
connected to the PSTN. It would look something like this (pseudo-code
follows):
if ($a){
dial(8005551212);
}else{
dial(866555);
}
The part
Hi,
Eric Bishop wrote:
thanks for that. Do you know what P-Asserted-Identity needs to be set to to
hide caller ID via privacy headers?
It must a URI (commonly a SIP URI) and may contain an optional
display-name. But it's Privacy header field that determines if asserted
identity information
Hello,
Jason Aarons (US) wrote:
I have a telco providing DTMF inband, they say they can't provide it any
other way. This is creating headaches for me.
What is the common method for SIP DTMF? Kpml, or 2833 or inband?
Isn't RFC 2833 method of transmitting DTMF tone information the same as
I have a few questions with regards to Digium S101I adapter.
I would like to use it as a traveling companion, plugging it into
various networks (all behind firewall I assume).
I'll be registering it into my asterisk server (behind firewall, port
4569 will be open).
1.) If I plug that small
Hi Roy,
Look I dont know why u specify 'zap/1-1', but i do things like this on
my agi scripts a lot of times:
...
$stdin= fopen('php://stdin', 'r');
$stdout = fopen('php://stdout', 'w');
$stdlog = fopen('/tmp/outPUT.log', 'a');
...
fwrite($stdout,EXEC DIAL
59 matches
Mail list logo
