Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys
sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so
it's very likely a Sipura. What's wrong with these people? :)
ftp://ftp.linksys.com/datasheet/pap2_ds.pdf
ftp://ftp.linksys.com/pdf/pap2_ug.pdf
Odd, I have absolutely *zero* issues with Promise PATA cards... I use
strictly software RAID on both SCSI and IDE on Linux 2.4. Never had
issues
with the kernel failing due to I/O load on rebuild or dealing with failed
drives.
Note: You can easily throttle the I/O bandwidth used for
Jay Milk [EMAIL PROTECTED] lazily top-posted:
Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys
sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so
it's very likely a Sipura. What's wrong with these people? :)
Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
Than!
Roland Zagler
mailto:[EMAIL PROTECTED]
@fog smart partners
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[EMAIL PROTECTED]
On Sun, 2004-08-22 at 07:35, Roland Zagler wrote:
Please can someone send me the .tar.gz file of spandsp, the site is
offline and i didn't find it anywhere!
http://sremington.zapto.org/downloads/asterisk/spandsp/ until the
opencall.org DNS servers are back up.
-Seth
--
Seth Remington
On Fri, 20 Aug 2004, Robert Boardman wrote:
BT are providing a SIP gateway for PSTN through the BT communicator with
Yahoo Messenger, I have done an ethereal trace and found that the BT
Communicator side of the software is using SIP, so in theory I could add
more PSTN lines to Asterisk for
Hi there,
app_mp3 still does not work with the latest bri-stuff patch and the
zaphfc driver. Here in my place it only works with the patch attached.
For me it seems the bri-stuff worsens the asterisk timing... has anybody
else made experiences with it?
Deti
Index: app_mp3.c
At 11:45 AM +0100 on 8/22/04, Kevin Walsh wrote:
Jay Milk [EMAIL PROTECTED] lazily top-posted:
Linksys is in bed with Cisco, Cisco has the (lousy) ATA-186, and Linksys
sells the PAP2-NA -- yet the user-guide DOES show Sipura commands, so
it's very likely a Sipura. What's wrong with these
I have an Adtran tsu120 with an FXO card in it. Can this be used for PSTN to Asterisk?
If so, do I use a serial v.35 cable? I do not have one.
Thanks
John Hill
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[EMAIL PROTECTED]
Hi
(B
(BRelease 1.0 of the FWD Assistant for MacOSX is now
(Bavailable for download.
(B
(BOne bug found in pre-release 1 had already been fixed in
(Bpre-release 2. Release 1.0 further adds automatic
(Binstallation of FWD's public keys in
(B/var/lib/asterisk/keys if they don't already exist
Since the largest regional ILEC in Brazil completely stopped taking new BRI
ISDN customers and started heavily incentiving their existing customers to
migrate over to ADSL (regardless if what they need is voice or data), we have
thousands of ISDN phones collecting dust.
Is there any network
I ran it like 10 times just in case:
[EMAIL PROTECTED] asterisk]# ztcfg -vv
Zaptel Configuration
==
Hi, please tell me,
is original skinny support in Asterisk stil under development or is better to try
chan_sccp from
http://chan-sccp.sourceforge.net ?
my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during
Asterisk startup)
and my phone (C7940) seems to be not
On Sun, Aug 22, 2004 at 06:11:10PM +0200, Pavel Jezek arranged a set of bits into the
following:
Hi, please tell me,
is original skinny support in Asterisk stil under development or is better to try
chan_sccp from
http://chan-sccp.sourceforge.net ?
my first try was unsuccessfull (chan_sccp
On Thu, Aug 19, 2004 at 11:25:17AM -0500, Rob Fugina wrote:
I'm using the latest Asterisk from CVS, SpanDSP 0.0.1k, and the
latest
app_rxfax.c (as mirrored by friendly list members recently), and
libtiff
3.5.7. Asterisk is detecting the fax signal properly, and
executing
the fax extension
Hi,
The incomming call is coming, asterisk answers.
So far 1 sip channel is used. The person who is calling choses either 1 or
2 and asterisk uses second channel to make a phone call.
If everything is ok two parties are talking to each other.
Now, the third person is calling, and all I want is
On Sat, Aug 21, 2004 at 08:35:22AM -0500, Lyle Giese wrote:
This doesn't answer your question completely, but I have noticed that
inserting and removing the kernal modules doesn't work all that well and
that rebooting is a better answer at that point.
Okay, I'll stop trying that, then. :)
[EMAIL PROTECTED] wrote:
Hi,
The incomming call is coming, asterisk answers.
So far 1 sip channel is used. The person who is calling choses either
1 or 2 and asterisk uses second channel to make a phone call.
If everything is ok two parties are talking to each other.
Now, the third
I am trying to compile the meetme2 application with the latest CVS head and
it fails. Here is the error message that I get. Can someone point me in
the right direction?
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include
Hi,
I had music on hold working but now don't know what happened.
I get :
WARNING[...] res_musiconhold.c:334 moh0_exec: Unable to start music on hold
(class '') on channel SIP...
Any ideas what is wrong ?
Regards,
Robert.
___
Asterisk-Users mailing
About faxing, anyone tried this?
How about instead of sending the fax to a tiff file, dial an FXS
extension, where you'll have either a dedicated fax machine, or a
Hylafax-connected modem, so you handle all your faxes though hylafax?
I have hylafax running succesfully, and can send and receive
Tim // NCS wrote:
Im new to this list, and ran across this post about a Uniden UIP200. Since its
been a few months now, I was wondering how it's turned out so far.
Tim,
We have deployed several UIP200 phones (22 to be exact).
The phone hardware is of exceptional quality, and it contains some very
If you are going to do hylafax why not just do it seperate from
asterisk on a regular modem and just email o ut the results. Don't see
the big bonus to using a FXS and the adding cost and point of
failures.
On Mon, 23 Aug 2004 11:13:25 +1200, Carlos Hernandez
[EMAIL PROTECTED] wrote:
About
[EMAIL PROTECTED] wrote:
I am trying to compile the meetme2 application with the
latest CVS head and
it fails. Here is the error message that I get. Can someone
point me in
the right direction?
gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g
I only suggested that in case someone has managed to receive faxes from
other remote asterisk machines, or via SIP...?
Otherwise, you're right.
Carlos
William Suffill wrote:
If you are going to do hylafax why not just do it seperate from
asterisk on a regular modem and just email o ut the
Hi there,
I've got an installation where there's 12 POTS line incoming into *, and
am trying to get some insight as to which VoIP hard phone would be most
suitable for this scenario.
Most of the VoIP phones I've looked at only have 4-6 line presentations;
is anyone aware of one that has more?
Hi,
I am using * to guide my callers throught my company's support menu. But
I have problem when the caller has a pulse dial telephony. Could *
detect digits dialed on pulse telephones?
Daniel
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[EMAIL PROTECTED]
Title: Message
I am writing a call
center application.
I do not want to use
Queues to manage my incoming calls and connect them to the operators for a few
reasons which I wont go into here.
The option I come up with is to create a context that the call
goes to which runs background() and
On Sun, 22 Aug 2004 23:07:04 -0300, Daniel Bichara
[EMAIL PROTECTED] wrote:
Hi,
I am using * to guide my callers throught my company's support menu. But
I have problem when the caller has a pulse dial telephony. Could *
detect digits dialed on pulse telephones?
I don't know of any IVR
On Sat, 2004-08-21 at 16:37, Paul Concepcion wrote:
well, that's our setup (8 analog lines - channel bank - t100P), so
it looks like DNIS is out of the question. We do have 8 phone numbers
though. Could we have a 1-800 number direct to each of those, then do
what you suggested with contexts?
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would be
most suitable for this scenario.
What would you guys recommend?
A Cisco 7960 with the 7914 expansion module [
Jeremy Bogan wrote:
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would be
most suitable for this scenario.
What would you guys recommend?
A Cisco 7960 with the 7914 expansion module [
The link work fine, the link is
http://www.ict.tuwien.ac.at/staff/darilion/ActXPhone/
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Robert
Rozman
Enviado el: Sábado, 21 de Agosto de 2004 04:11 p.m.
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users]
el Flynn wrote:
Hi there,
I've got an installation where there's 12 POTS line incoming into *,
and am trying to get some insight as to which VoIP hard phone would
be most suitable for this scenario.
Other than the incoming lines, the receptionist would need the normal
keyphone type stuff
On Mon, 2004-08-23 at 13:02, [EMAIL PROTECTED] wrote:
I am writing a call center application.
I do not want to use Queues to manage my incoming calls and connect
them to the operators for a few reasons which I wont go into here.
It would be interesting to see why this wouldn't work for you
Hi,
in response to a previous posting regarding getting the x100p to
work, I was told to run zap show channels, but when i do i get no
such command 'zap'
There was a previous posting on this, but the guy never posted the solution.
thanks,
Imran
___
Imran Akbar wrote:
Hi,
in response to a previous posting regarding getting the x100p to
work, I was told to run zap show channels, but when i do i get no
such command 'zap'
There was a previous posting on this, but the guy never posted the
solution.
chan_zap.so is not loaded.
Jeremy
Imran Akbar wrote:
Hi,
in response to a previous posting regarding getting the x100p to
work, I was told to run zap show channels, but when i do i get no
such command 'zap'
There was a previous posting on this, but the guy never posted the
solution.
you might want to double-check what you
Thanks Jeremy,
but how exactly do I load chan_zap.so? I put it into my
modules.conf, but when i run asterisk now it says it can't find it
(loading module zap_chan.so failed). It doesn't seem to be on my system...
thanks
imran
Jeremy McNamara wrote:
Imran Akbar wrote:
Hi,
in response to
Imran Akbar wrote:
Thanks Jeremy,
but how exactly do I load chan_zap.so? I put it into my
modules.conf, but when i run asterisk now it says it can't find it
(loading module zap_chan.so failed). It doesn't seem to be on my system...
Have you compiled, installed and configured Zaptel?
to the best of my knowledge, i have, but i'm redoing it. i'm looking at
the instructions at
http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation
is that the best guide?
thanks
Imran
Jeremy McNamara wrote:
Imran Akbar wrote:
Thanks Jeremy,
but how exactly do I load chan_zap.so? I put
It should be chan_zap.so not zap_chan.so.
Imran Akbar wrote:
Thanks Jeremy,
but how exactly do I load chan_zap.so? I put it into my
modules.conf, but when i run asterisk now it says it can't find it
(loading module zap_chan.so failed). It doesn't seem to be on my
system...
sorry, my bad. typo in the email, but it was correct in modules.conf.
Im trying to reinstall the zaptel stuff, but i'm not seeing anything in
var/log/messages after doing my modprobe's?
Thanks
Jon Radon wrote:
It should be chan_zap.so not zap_chan.so.
Imran Akbar wrote:
Thanks
Imran Akbar wrote:
sorry, my bad. typo in the email, but it was correct in modules.conf.
Im trying to reinstall the zaptel stuff, but i'm not seeing anything in
var/log/messages after doing my modprobe's?
Thanks
try running the dmesg command - the digium stuff appears there instead
of
Thanks,
I seem to have done the zaptel installation - what am I missing - i
still don't have a chan_zap.so file?
in my zaptel directory:
make clean
make
make install
modprobe zaptel
modprobe wcfxo
got stuff in dmesg
did a make config in the zaptel directory
edited the zaptel.conf,
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