Steven J. Sobol wrote:
X-Lite build 1079 consistently chokes no matter which codec I use -
after five seconds I suddenly have no sound coming in and possibly no
sound going out too. Putting the line I'm on on hold and then switching
back to it gives me another five seconds of sound, then it
On Tuesday 14 October 2003 23:50, Chris Albertson wrote:
If the software needs a specialcard to keep time then the
software is broken or poorly designed.
Don't complain so loudly unless you're willing to contribute the
fixes. Opinions are like assholes, and you know where that's going.
Takes
Steven J. Sobol wrote:
On Wed, 15 Oct 2003, Jon Pounder wrote:
Nothing works. Call transfer and call waiting, in particular. (Well,
almost nothing; vm notification does work)
Call transfer and call waiting do work, although the call waiting is a
little loud and anoyoing.. :)
There is no
Our experience with the Budget Tones 101have been poor as well. The
devices seem to die after a day or two (even with the new firmware) and
then need to be rebooted. On occasion, the device needs to be literally
unplugged and plugged back in as even the reset doesn't work.
There are some nice
On Wed, Oct 15, 2003 at 12:50:33PM -0500, Stephen Dredge wrote:
I've seen this question asked before but haven't seen a definative answer.
Does the e100p work in australia? Did any one who was asking the question
You need written permission from your Telco to use non-approved hardware,
I
Dear Gus,
Thanks for the info. I am quite sure I have some Siemens hardware on the
other end of our E1 ... what do you think I should have as pridialplan
in my zapata.conf ?
Thanks
On Wed, 2003-10-15 at 01:49, CW_ASN wrote:
Marcel:
Some switches with particular functionalities don't expect
Andrew Joakimsen wrote:
Has anyone tested using SIP endpoints (Possibly the ATA-186) with a
connection that has at least 200ms, if not more, of latency? We are
trying to get some stuff setup in Australia and wanted to know if this
would be feasable, are there any added delays? Echos?
I am
I'm a pretty big fan of OpenBSD for various reasons, and I have been itching to get
Asterisk running on that platform, since it is what the rest of my network has as it's
base operating system. I have apparently been asleep at the wheel, since I didn't
hear that Asterisk is now successfully
Paul Cheng wrote:
Our experience with the Budget Tones 101have been poor as well. The
devices seem to die after a day or two (even with the new firmware) and
then need to be rebooted. On occasion, the device needs to be literally
unplugged and plugged back in as even the reset doesn't work.
Due to some failed hardware on another platform, I've had to move a
T100P card to a different chassis. After this move was completed, I
am seeing some strange results on the T100P card that do not display
to me any failure mode with which I am familiar. The card comes up,
and shows good
John Todd wrote:
Due to some failed hardware on another platform, I've had to move a
T100P card to a different chassis. After this move was completed, I
am seeing some strange results on the T100P card that do not display
to me any failure mode with which I am familiar. The card comes up,
On Wed, 2003-10-15 at 10:16, Brian Capouch wrote:
Mine is a small sample set (~10) but a couple of them are the primary
phones I've been using--I'm a pretty heavy user--and over several weeks
we seem to be doing fine with them. The CW tone is extremely annoying,
and one has to be a bit
Due to some failed hardware on another platform, I've had to move a
T100P card to a different chassis. After this move was completed, I
am seeing some strange results on the T100P card that do not display
to me any failure mode with which I am familiar. The card comes up,
and shows good
It works with a third party gatekeeper, which is good enough (get asterisk
to register with a gatekeeper). There's 20 lines in asterisk where Jeremy
started making the gatekeeper functionally, currently rapped with #if 0
#endif I doubt it will ever exist, for the short term anyway.
-
On Tue, 2003-10-14 at 22:15, Brian Schrock wrote:
Hello,
I resolved my echo issue using grandstream/estara etc etc sip phones and
wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl
motherboard for an asus p4pe? i845? based motherboard and the echo has
completly gone
hi!
Is there acdr analyzer for
* to extract the call details from mysql cdr ?
Here's the scenario I wan't to
analyze.
1. internal user A - operator
2. operator - make outside calls
3. operator connect the above 2
parties.
I get a log for step 1 and 2. But not for the 3rd
step.How do I
Hi,
At 17:14 14-10-2003 -0500, you wrote:
I have Asterisk up and running and it is working great with my SIP phones.
However, I have some Skinny-protocol Cisco 7960s. Does Asterisk support
the Skinny protocol? I've seen some references to Skinny in the software.
If so, should I stick with
I know this topic has been done before but I cant find an answer for it
anywhere.
I have Grandstream phones running with my Asterisk server.
Is there anyway to do announced transfers with these phones?
i.e. A is talking to B, A then presses a key to initiate transfer, calls C,
tells C that B
This may or may not be the same problem that appeared on one of my E/T100P
cards a few months ago.
I got lots of timing errors and finally figured out that the capacitor C31
was broken (no visible damage, though). When I soldered an 1.0 uF capacitor
over it, the card started to work perfectly
[EMAIL PROTECTED] wrote:
I know this topic has been done before but I cant find an answer for it
anywhere.
I have Grandstream phones running with my Asterisk server.
Is there anyway to do announced transfers with these phones?
i.e. A is talking to B, A then presses a key to initiate transfer,
I only have 1 but the absolutly only time it has to be rebooted is when I
change a parameter or upgrade the firmware. It has run for weeks without
any problem. Another poster mentioned the 10 vs. 100 Ethernet speed.
Maybe Grandstream can upgrade the interface in future hardware. I don't
imagine
Hi everybody,
I'm trying to use MeetMe(2000|p) in order to enter in a conference
room. But when a mobile or a SIP call press '#' to go out from it,
everybody goes out. Instead when a analg press it, all works fine.
Anyone else have this problem?? Can anybody help me please??
Thanks a lot.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
How do I prevent call forward looping over multiple Asterisk systems?
I.e.
SIP1---Ast1===Ast2---SIP2
SIP1 forwards his calls to SIP2 and SIP2 forwards to SIP1.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
Alastair:
At least in some European countries and southamerican environments, you
can't send your own ANI.
For example: if you have a PRI with the number: 5288 to 52880099, you
could send as ANI any number between 000 and 099. The fixed part will be
transmitted by your PSTN switch to the
On Wed, 15 Oct 2003, WipeOut wrote:
[EMAIL PROTECTED] wrote:
I know this topic has been done before but I cant find an answer for it
anywhere.
I have Grandstream phones running with my Asterisk server.
Is there anyway to do announced transfers with these phones?
i.e. A is talking
Adam Hart wrote:
From: Anton Tinchev [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 1:06 PM
Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
For timing...
I'm first to buy 5 pack. Even for $30.
Doesn't ztdummy already do this?
On Wed, 15 Oct 2003, rnc Info Lists wrote:
One option I would definatly like is the ability to turn off the ringer.
Since my testing ususally happens after my wife goes to bed it would help
NOT to have the audible ring but only the visual indication!
Better still I would like volume control
Michael T Farnworth wrote:
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
Yes, I have found the Snom200 does consultative transfers well..
Couldn't this problem be solved with an asterisk upgrade?
No, Its an issue that is handled on the phone..
Michael,
That would work for me too. If the volume can be reduced (maybe to zero or
almost zero) then my request for the ability to disable it is not needed.
Since the volume of the speaker and handset can be controlled maybe the GS
folks can include a patch in the next release of the firmware
If the software needs a specialcard to keep time then the
software is broken or poorly designed.
Don't complain so loudly unless you're willing to contribute the
fixes. Opinions are like assholes, and you know where that's going.
Takes something else entirely to fix a perceived problem.
I resolved my echo issue using grandstream/estara etc etc sip phones and
wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl
motherboard for an asus p4pe? i845? based motherboard and the echo has
completly gone away along with aggressive suppressor option in the
makefile.
Anton Tinchev wrote:
Adam Hart wrote:
Sent: Wednesday, October 15, 2003 1:06 PM
Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
For timing...
Doesn't ztdummy already do this?
Only if you has the right usb chip
And the winner is? :-)
There's no comment in the source code
Marcel:
Generally, the ton for A subscriber is setted as National Number, and ton
for B subscriber is setted as local.
In the specific case for Siemens EWSD (wonderful switch), it does a
conversion a digit using an internal table, this tables do this:
EWSD Side
Hi,
a little newbie question:
I've just installed asterisk and played a little with it. the server has a
pubblic address while the clients (sjphone, msn messenger, sipset) are
behind a firewall/NAT. sip part always works, while rtp part sometimes
works, sometimes not. the question is: does
On Wed, 15 Oct 2003, WipeOut wrote:
Michael T Farnworth wrote:
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
Yes, I have found the Snom200 does consultative transfers well..
Couldn't this problem be solved with an asterisk upgrade?
Agressive Suppressor is active in the makefile.
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 8:07 AM
Subject: Re: [Asterisk-Users] WCFXO echo rexolved for me
I resolved my echo issue using grandstream/estara etc
rnc Info Lists wrote:
Its a free world and everyone is entitled to their opinion. Here's mine
on this topic. The cards aren't so expensive (99.95 USD). If they have
their own hardware then they don't have to depend on the target system
having a particular configuration. Example: right now I
Paul:
in your opinion, which hardware SIP phone is the best price/performance
device after taking into account support costs?
Regards,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Cheng
Sent: Wednesday, October 15, 2003 2:57 AM
To: [EMAIL
Michael T Farnworth wrote:
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
Yes, I have found the Snom200 does consultative transfers well..
Couldn't this problem be solved with an asterisk upgrade?
No, Its an issue that is
Michael T Farnworth wrote:
Perhaps I am confused, but I tend to believe that Asterisk sits in the
middle of all these calls. So when I press the # key for transfer it
could actually put the incoming call, allow me to dial and then speak to a
person and then when I press # again redirect that
This is troubling. Shouldn't your hubs/routers autosense the 10MBPS?
-- Original Message --
From: WipeOut [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Wed, 15 Oct 2003 07:53:13 +0100
Steven J. Sobol wrote:
On Wed, 15 Oct 2003, Jon Pounder wrote:
Where do I get one and how do I order it?
-Original Message-
From: Steve Meyers [mailto:[EMAIL PROTECTED]
Sent: Wed 10/15/2003 1:27 AM
To: Asterisk List
Cc:
Subject: Re: [Asterisk-Users] Wildcard TDM400P - FXO?
Most of the time, when someone calls in from the outside on a POTS line,
and possibly over IAX as well, they don't hear any ringing sound while
the internal SIP phones ring. If you call from an inside SIP phone,
even forcing it into the incoming context, you hear the ringing.
The outside
On 15 Oct 2003 00:34:55 -0300
Juan J. Sierralta P. [EMAIL PROTECTED] wrote:
I don´t have this kind of problem on my Cisco 7960 which has
VAD
deactivated. The problem I don't see any VAD option in AudioModes of
ATA.
--
Juanjo sin .sig
You can disable VAD seting the
Hello list,
I have some trouble with chan_capi and asterisk. I'm using 2 Fritz!Card PCI in one *
box and testing it with soft phone like sj-phone and x-lite without any serious
problems, but:
+ I don't understand some settings.
- what's deflect in capi.conf for?
- how to configure multiple
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 11:04 AM
Subject: Re: [Asterisk-Users] Announced Call Transfer
Michael T Farnworth wrote:
more expensive phone in reception but leave the other people on the
cheap
On Wed, 15 Oct 2003, Rich Adamson wrote:
Michael T Farnworth wrote:
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
Yes, I have found the Snom200 does consultative transfers well..
Couldn't this problem be solved with an
That is my thought too - I only have a couple cheap 4 port switches that
won't autosense, and are pure 100, everything else runs fine at either 10
or 100.
At 09:17 AM 10/15/2003, you wrote:
This is troubling. Shouldn't your hubs/routers autosense the 10MBPS?
-- Original Message
Its a free world and everyone is entitled to their opinion. Here's mine
on this topic. The cards aren't so expensive (99.95 USD). If they have
their own hardware then they don't have to depend on the target system
having a particular configuration. Example: right now I am running * on
a
Just a couple of notes on the ISDN issue in Australia...
I agree with the audio quality of the NETjet-s. I also had a horrible time
working out the DTMF detection under ISDN4Linux. Overall, I can not fault
the hardware and guys at Traverse were quick to help out but it is not
production quality
Which telephones support this requirement and is it possible to have a
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
The Cisco 7960s and 7940s (what we use) support this just fine.
You have two options on the soft keys when in a call Trnsfer and
Digium tells me that they are getting BETA boards in some time this week for
their internal testing. If everything checks out OK they should be out
soon.
Sean
- Original Message -
From: Gene Kochanowsky [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 9:19 AM
The only thing that is wrong is that there seems to be some expectation of
Digium that they have to tell things... The source code is available. If
someone isn't happy with the Digium methods then they should find a
solution and post it to the list and/or one of the several Asterisk Wiki's
that
Ever since late September I am unable to get any newer version of zaptel drivers
on a machine I keep at home to play with.
If I am not home and the answering machine gets the call the system hangs up
on the caller during the greeting but ifI'm there to answer it, the call will work fine.
If I
On Wed, 2003-10-15 at 09:35, Eduardo Goncalves wrote:
I dont have this kind of problem on my Cisco 7960 which has
VAD
deactivated. The problem I don't see any VAD option in AudioModes of
ATA.
--
Juanjo sin .sig
You can disable VAD seting the AudioMode bit 0.
On Wed, 2003-10-15 at 09:20, Olle E. Johansson wrote:
Anton Tinchev wrote:
Adam Hart wrote:
Sent: Wednesday, October 15, 2003 1:06 PM
Subject: [Asterisk-Users] Digium should develop and sell just Dummy card.
For timing...
Doesn't ztdummy already do this?
Only if you has the right usb
Skinny is still a little less mature then SIP, so if you want a production
solution, go with SIP for now. If you have some time to help
develop/debug,
stick with Skinny :-)
I do. I'll work on it.
You can probably find SIP loads for the phone around somewhere, although I
don't have them
Hello,
Do you know a **real** sip fax server to _send_ faxes with Asterisk?.
By example: an employee sends a TIFF facsimile by email to the SIP fax server;
SIP fax server uses SIP to communicate with Asterisk; Asterisk communicates
with a Cisco SIP2PSTN Gateway via SIP; Cisco sends the fax to
I was using the Asterisk::Manager perl module and had some troube with it
so I decided to make my own. I have a pretty good prototype after a few hours
(My main point about the protocol is at the bottom of this example.)
http://asterisk.650dialup.comis where you can download it.
my $man = init
Hi list,
I have a cisco 827 with 4 fxs and an * gateway, like this:
[c827]--sip-[asterisk]-em---PSTN
The codec used is g711alaw over a 9Mb link.
Some calls just hang up after some minutes of conversation. Cisco shows
a DisconnectText=normal call clearing
A nice flag in app_dial ?
would be f and F to indicateweather to send flash to
the fxo or fxs device in a bridged call.
so if youhave a pots line on an x100p bridged to a tdm400p
and the pots line has call waiting you hear the call waiting signal
'f' would behave as normal and flash on the
On Wed, 2003-10-15 at 09:17, Andrew Kohlsmith wrote:
Its a free world and everyone is entitled to their opinion. Here's mine
on this topic. The cards aren't so expensive (99.95 USD). If they have
their own hardware then they don't have to depend on the target system
having a particular
Since you guys are on this topic already..
We had a Braxtel switch and a call would come in and then if you needed
to transfer the caller to someone outside the office to a number
unrelated to our company we would have to tie up two of our lines in our
switch while their conversation took place.
set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf
On Wed, 2003-10-15 at 09:50, Eduardo Goncalves wrote:
Hi list,
I have a cisco 827 with 4 fxs and an * gateway, like this:
[c827]--sip-[asterisk]-em---PSTN
The codec used is g711alaw over a 9Mb
on 10/12/03 10:08 PM, Philipp von Klitzing at
[EMAIL PROTECTED] wrote:
You might also want to take a look at the Swissvoice products
(www.swissvoice.net). As far as I have been able to judge from the web
site those phones offer considerably more for the same price.
Next to that I'd be very
more expensive phone in reception but leave the other people on the cheap
Grandstream phones?
Yes, I have found the Snom200 does consultative transfers well..
Couldn't this problem be solved with an asterisk upgrade?
No, Its an issue that is handled on the phone..
Hi!
I have three questions:
- I have called from an H.323 softphone to a SIP one, and then I have tryied to
transfer the call to be accepted by asterisk. And it has not work. Is it
possible to do that? And if it is possible, what I have to do for that works?
- In extensions.conf, there's the
Are there USB PCI cards that use the chip that's compatible with
ztdummy?
So now you're using a $50 USB PCI card, or a $100 FXO/FXS card... you're
adding stuff to the system either way. :-)
Personally I really like the RTC dummy driver. It just locks the alarm at
1024Hz and calls the
One for the gurus..
I have seen there has been a lot of discussion about using SER with
Asterisk.. This to me seemed like an over kill becasue it would
basically be doing most of what Asterisk is doing anyway unless you
create some weird and wonderful config in SER..
Anyway, I decided to go
On Wed, 2003-10-15 at 11:56, Andrew Kohlsmith wrote:
Some one else here has mentioned the quality of software design due to
the need for hardware timing. This should be addressed by the fact that
many tools are using hardware timing. Mp3 players use the sound device
as a timing source.
Was I that particular poster? Maybe. But yes! there is
absolutly no need for specialized hardware to meter an 8Ksps
stream out of a PC.
Your example of a MP3 player depending on the hardware is valid but
what about a video player? What does it use to keep a constant
30FPS rate?
Now
Steven,
Good comments but remember good enginerring starts with reading
the requirements and desiging to those requirements.
in the case of SIP at least these is an RFC. What is the
timming requirement on media packets? How is the stream
synchronized? I'll read it in the next few days but
Hi!
I am doing a research about the prices of SIP telephones. If someone can tell me
which one are the cheapest and have an acceptable quality... it will be very
kind.
Best Regards,
Mireia
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Why make additional hardware whose driver needs be ported to other
systems when you can make similar dummy interfaces? What happens to the
systems that don't have PCI buses. I can think of some older MAC
hardware that can run linux, but not PCI cards. Not that I would bother
my self with one
On Wed, 2003-10-15 at 12:50, Chris Albertson wrote:
Steven,
Good comments but remember good enginerring starts with reading
the requirements and desiging to those requirements.
in the case of SIP at least these is an RFC. What is the
timming requirement on media packets? How is the
Andrew,
I am running it rather well on a original Pentium 100 Mhz, 32 MB RAM, no
USB adapter. I agree with you this would not be an ideal setup for a
business but in a home it will work rather well. I think it'll handle 2 CO
analog lines fine.
Yes, my wife thinks its overkill. Probably is, but
Chris,
Good point. As I understand it, the Asterisk software requirement was to
be a PBX between normal telephone lines and VoIP. Maybe even it was just
to replace the expensive PBXs. As such seems to me that it clearly met
and exceeded its design requirements since it utilizes the hardware
Hi All,
I am kind of new to asterisk. Here is a little prolem that I am facing.
Here is my problem and questions: I am just adding two gnophone users to my dialplan, all three systems are within lan.
1. in iax.conf:[mako] type=friend auth=pliantext secret=myown context=default host=dynamic
On Wed, 15 Oct 2003, M.A. Ali wrote:
I am kind of new to asterisk. Here is a little prolem that I am facing.
Here is my problem and questions: I am just adding two gnophone users to
my dialplan, all three systems are within lan.
1. in iax.conf:
[mako]
type=friend
auth=pliantext
Lets see if I understand this logic. I'll restate it:
1) Asterisk's MOH is only broken if you attempt to build a
VOIP-only system
2) Asterisk is not intended for such use. It is a PSTN
oriented PBX that just happens to handle VOIP.
3) Therefore Asterisk is not broken
OK. If you believe
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference room the following message is played:
That is not a valid
ok,
forget the homemade ID thingI found ActionID actually does it
but i still mean the part about every item having headers etc
Do you Yahoo!?
The New Yahoo! Shopping - with improved product search
On Wed, 2003-10-15 at 13:57, Chris Albertson wrote:
Lets see if I understand this logic. I'll restate it:
1) Asterisk's MOH is only broken if you attempt to build a
VOIP-only system
Supposedly this was fixed recently. As I don't use MOH, nor am I ever in
a VoIP only system I can't
Hi all:
I've got some core dumps with chan_skinny. The
client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is
CVS-10/05/03-16:03:26.When I make a call, the phone connected with ATA rings
only 1 time and * dies. Maybe I have some errores in ATA config. If someone has
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
On Wed, 15 Oct 2003 11:16:03 -0500
Eric Wieling [EMAIL PROTECTED] wrote:
set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf
Thanks for the tip. Could you explain me why these options set to yes
may cause the hang up?
At this time, I don't have these options in
On Wed, 2003-10-15 at 13:57, Andrew Kohlsmith wrote:
Are there USB PCI cards that use the chip that's compatible with
ztdummy?
So now you're using a $50 USB PCI card, or a $100 FXO/FXS card... you're
adding stuff to the system either way. :-)
Personally I really like the RTC dummy
Hi, I´m trying to make * work with Brazilian analog signalling..
I´m using the following in indications.conf file...
[br]
description = Brasil
ringcadence = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/4000
callwaiting = 425/60,0/250,425/60,0/5000
I changed zaptel.conf to
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to
On Wed, 2003-10-15 at 11:46, Juan J. Sierralta P. wrote:
You can disable VAD seting the AudioMode bit 0.
Thanks Eduardo but I have Silence suppression off :(
The strange thing its that even that MOH is running against the ATA
the stats on the channel (show channel SIP)
The default should be no. Both options listen to the audio stream.
busydetect tries to determine if it hears a busy signal and if so
disconnects the call. callprogress tries to determine if the call has
been disconnected and disconnects the other legs of the call. Both
options are buggy cause
On Wed, 2003-10-15 at 14:48, rnc Info Lists wrote:
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such
device
Hint: insmod errors can be caused by
Take a look at zaptel/zonedata.c, I guess you have to change it.
Greetings from Rio de Janeiro ;-)
PHM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Lomonaco
Sent: quarta-feira, 15 de outubro de 2003 16:40
To: '[EMAIL PROTECTED]'
Subject:
There was a tread that I googled for and could not find
about Asterisk being open to SIP DOS Attacks. I have a customer whose machine
was hammered last light by traffic on its SIP port causing the OS to use up its
resources. Namely number of open files. The discussion was around the fact
Hi,
I have Grandstream 101 and I'm OK with... The cost is about 70 USD.
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mireia Munoz
de jesus
Sent: Wednesday, October 15, 2003 2:01 PM
To: [EMAIL PROTECTED]
Subject:
Asterisk will bridge a call in some cases and not in others. If codec
conversion is required between phones, its stays in the middle. If the
two phones can agree upon a common codec, etc, * is not in the middle
from a pure communications perpective. In that particular case, what
the phone does
On Wed, 2003-10-15 at 15:10, rnc Info Lists wrote:
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No
On Wed, 15 Oct 2003, WipeOut wrote:
Steven J. Sobol wrote:
X-Lite build 1079 consistently chokes no matter which codec I use -
after five seconds I suddenly have no sound coming in and possibly no
sound going out too. Putting the line I'm on on hold and then switching
back to it gives me
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