compiles.
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Tony
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a make in asterisk-addons.
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still
make outgoing calls.
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HT286, and it seems to behave much
better.
I have two BT102 phones running 1.0.4.68, and one of them still does it.
Register Expiration is set to 3 minutes.
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and it sounds like one or two 3GHz CPUs should do it, but if anyone
reading has direct experience of this kind of application, I'd be
very grateful for any comments.
Thanks
Tony
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don't you want to use sox? I see from http://sox.sourceforge.net/
that it is available for Windows, and I would expect that as it compiles
for BSD it would also compile for Mac OSX.
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Tony
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messy
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sure makering.pl has the x bit turned on, wherever
the file is located.
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the PostgreSQL sections,
and it looked like you also didn't have zaptel installed.
Cheers
Tony
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must be generated as ring1.bin
and not just renamed from another file name. I haven't tested this
theory.
You then need to go to the phone's web page to tell it to use the new
ringtone.
Cheers
Tony
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somewhere visible?
Cheers
Tony
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store.yahoo.com or www.digium.com
you will be helping to support Digium, who gave Asterisk PBX to the
community. This is a Good Thing (TM).
The other three are OEM copies of the same card, but do not benefit
Digium at all.
Cheers,
Tony
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In article [EMAIL PROTECTED],
Leo Ann Boon [EMAIL PROTECTED] wrote:
The new TDM400P with FXO doesn't take up any IRQ. I've 2 boards and both
are not using any IRQ.
Weird - does that mean they can't provide Zaptel timing like the X100P can?
Cheers
Tony
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, work fine , just
chunk error message
checksum before = db8e
checksum after = 4db2
checksum failed
Are you running the perl program on Unix/Linux or on Windows?
It has only been tested on Linux, and may need binmode STDIN;
if running under Windows.
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with the extra 3 bytes, and another with the extra 1 byte?
There may be something my program hasn't taken account of, and it would
help me to find out what it is.
Thanks
Tony
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are no bigger than 65536 bytes.
Earlier versions of my program didn't check for this, but the latest
one does.
Cheers,
Tony
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Which is the correct fax number for disclaimers?
http://bugs.digium.com/main_page.php says +1-256-864-0464
http://www.digium.com/bugtracker.html says +1-256-971-6890
Or are they both equally good?
Cheers
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it expects the kernel source tree to match the running kernel.
If you had built a new kernel called 2.6.5-1.315custom and then booted from
it, you would probably have built zaptel successfully. I think :-)
Cheers
Tony
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that can't.
Cheers,
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In article [EMAIL PROTECTED],
Fabio Donaggio [EMAIL PROTECTED] wrote:
Hi to all!!
Is there another method to download asterisk addons???
Another method in addition to what?
Tony
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into /tftpboot and rebooting my phone,
amazingly enough it works! I now have a new ringtone.
Time for bed
Cheers,
Tony
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the following:
Standalone SIP / IAX mode:
If you want to use Firefly on our Firefly phone network (with your own
voicemail etc.) then you will need to register a phone number. However,
you can also use Firefly as a SIP or IAX client on your own network.
Cheers
Tony
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in their FAQ they still explicitly say it
can be used with Asterisk systems.
Cheers
Tony
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, chan_zap and the zaptel driver. I've posted a couple of
emails asking questions to this or the asterisk-dev list over the last
few days, but have had no responses. Either no-one's interested in
answering, or no-one knows the answers!
Cheers,
Tony
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phones is set to canreinvite=no in sip.conf.
There may still be other reasons why it might not work, but if the phones
have done a reinvite, they are then sending their audio straight to each
other without it going through Asterisk. That would prevent recording.
Cheers
Tony
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In article [EMAIL PROTECTED],
Dave Packham [EMAIL PROTECTED] wrote:
have you tried the #asterisk-dev IRC room? thats the best place
Yes, I did, without much success. Perhaps I chose a quiet time.
Cheers,
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In article [EMAIL PROTECTED], Dan [EMAIL PROTECTED] wrote:
A new version with some cool features (not available on any other soft
phone) will be available at the end of the week.
Send me a mail if you need further assistance.
Will it support any codecs other than GSM?
Cheers
Tony
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, but not as quickly.
Comments?
Cheers
Tony
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!
Tony
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version..
http://www.voiptalk.org/products/gt_update.php
Cheers
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with. In
the case of fedora, you need to use gcc32. So the make command is:
make HOSTCC=gcc32
Hope this all helps!
Tony
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to a licence fee.
I believe G.726 can be used between Grandstream and Asterisk, but haven't
yet tried it.
Grandstream will soon be releasing firmware that supports ILBC.
(Anyone seen it yet? 1.0.4.55 doesn't appear to have it)
Cheers,
Tony
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and re-install the kernel sources, and make sure the link
/usr/src/linux2.4 points to the correct source tree that matches the
running kernel.
Cheers,
Tony
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In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
How can I remove callerid functionality?
That was mentioned on this list only a couple of days ago, and will be
in the mailing list archives.
In zapata.conf you need to include the line usecallerid=no.
Cheers,
Tony
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In article [EMAIL PROTECTED],
Bill McCready [EMAIL PROTECTED] wrote:
Where may I find a Windows driver for a Wildcard FXO Card ???
Why would anyone want such a thing?
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with this feature yet, so the above probably highlights both
(a) my lack of understanding, and (b) the lack of documentation!
Cheers,
Tony
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In article [EMAIL PROTECTED],
Tilghman Lesher [EMAIL PROTECTED] wrote:
On Thursday 15 April 2004 03:01, Tony Mountifield wrote:
This all seems rather cumbersome, and I haven't had the chance to
experiment with this feature yet, so the above probably highlights
both (a) my lack
requires a zaptel timing source. If you do not have any
zaptel cards in your system, you will need to install either ztdummy
(only if you have a uhci type of USB) or zaprtc.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
Cheers,
Tony
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see that the part which said
a zaptel timer was necessary for MoH was deleted on 3 Apr.
Is it correct that MoH doesn't need a zaptel timer? If so, was this
always the case, or did it change at some point?
Cheers,
Tony
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Hi, could anyone explain the intent and usage of the new e and E flags
in the MeetMe app? The Wiki doesn't mention them yet, and I have not
been able to find any other documentation of them.
Thanks
Tony
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In article [EMAIL PROTECTED],
Fran Boon [EMAIL PROTECTED] wrote:
On Mon, 2004-04-12 at 17:37, Tony Mountifield wrote:
The zaprtc.c code is based on the rtc.c from 2.4.20. I am running 2.4.22,
so I isolated the zaprtc changes, and re-applied them to a copy of the
rtc.c from 2.4.22. It works
, and it
would be used for nothing other than the clock, since there are no
other interfaces available for me to plug into the card. This was just
to nip the why don't you just pony up for a Digium card? responses :-)
:-)
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Tony
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of Bison solves the problem..
What kind of problems? I didn't notice any problems with building
Asterisk on FC1 with the supplied bison, but perhaps I wasn't paying
attention or didn't know what to look for.
Cheers
Tony
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installed? MOH needs the timer that they
provide. If not, and if your USB controller is the UHCI type, you
can use ztdummy instead. Failing that, you need to use zaprtc, which
is a modified rtc module. See www.voip-info.org for details.
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Tony
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-info.org - it's worth studying.
Cheers,
Tony
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that was made: make sure silence suppression
is turned off on your phone. Asterisk uses the incoming audio stream for
timing the outgoing one - if your phone turns off the stream it is sending,
it will stop the stream coming back too.
If it's not that, I'm out of ideas! :-)
Cheers,
Tony
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).
Can anyone suggest what things I should check or change?
Cheers
Tony
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fine).
Can anyone suggest what things I should check or change?
Cheers
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.
Create an account for yourself on www.voip-info.org and add it there.
Cheers,
Tony
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thing I have to try. Hopefully this evening.
Interested to see you are just up the road: I'm in Winchester.
Cheers,
Tony
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In article [EMAIL PROTECTED],
Eric Wieling [EMAIL PROTECTED] wrote:
On Wed, 2004-03-24 at 08:02, Tony Mountifield wrote:
In article
[EMAIL PROTECTED],
Robinson Tim-W10277 [EMAIL PROTECTED] wrote:
Sorry, didn't read your mail thoroughly - you've already tried
canreinvite
is installed. It's only kernel modules
that need to be compiled with gcc32, but there's no harm in trying it
for applications too, as, far as I know.
Cheers,
Tony
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of package X, you need to have package X-devel installed.
Cheers
Tony
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kernel modules (zaptel, zaprtc)
should also be built on FC1 using gcc32, not just cc or gcc. This
also gets rid of all those type punning warnings.
Cheers,
Tony
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to send DTMF via via RTP (RFC2833) with a payload
type of 101. (tried 100 and 102)
The first 2 codecs are set to PCMU and PCMA (tried to switch those
arround too).
Put dtmfmode=info in your sip.conf, and set the phone to use SIP INFO.
Then it will work.
Cheers,
Tony
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fine).
Can anyone suggest what things I should check or change?
Cheers
Tony
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with your problem though.
Cheers
Tony
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haven't found MoH to give any trouble. Do you have a Zaptel card
in your system? Music on Hold needs a timer, which is normally provided
by the zaptel driver. If not, you will need to use ztdummy or zaprtc.
See http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer
Cheers,
Tony
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Can anyone advise from experience what size of PC would be needed
to support two TE405P 4xE1 cards to provide conference bridging
for up to 20 concurrent conferences of 10 participants each?
All the participants would be on the E1 trunks, not VoIP.
Thanks in advance,
Tony
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