.
BTW: your extensions.conf looks silly, you'll only be able to call test3 from
test3.
Busy most of the time ;-)
Stefan Märkle
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OR you can try this:
in features.conf:
[applicationmap]
zapflash=*3,callee,flash
if you put any spaces in the above line, it will not work!!!
in extensions.conf add this line right before the dial commands where
you want this to work:
exten = s,12, set(DYNAMIC_FEATURES=zapflash)
and
flash-key, mobile phone called in via another zap channel = no difference
between the incomings
Has somebody any hints for me?
Stefan
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Stefan Märkle Netpioneer GmbH
Leitender Systemarchitekt Beiertheimer Allee 18
[EMAIL PROTECTED] 76137 Karlsruhe
there are means to transfer a SIP=IAX call to the
slave server where it would be replaced by SIP=ZAP termination but that would
Certainly be the most effective solution, since these slaves could be handled
Dynamically for scaling the installation.
Just my 2 cents
Stefan Märkle
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Stefan Märkle
change firewalls.
Any one has already done this? Any feedback will be much appreciated.
We're working on similar problems, so if you come up with a perfect solution,
please let me know.
Also, if you are interested in a commercial solution feel free to contact me
off-list.
Stefan Märkle
.
Greets
Stefan
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[EMAIL PROTECTED] 76137 Karlsruhe
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said: nice piece of work and at the moment the softphone of my choice on
W2k instead of firefly or purtel iaxphone I used before.
Have a nice day and greets from Germany
Stefan
--
Stefan Märkle Netpioneer GmbH
Leitender Systemarchitekt Beiertheimer Allee 18
mangled. Does your service script run
as the same user as when you try starting in a normal shell? Check permissions
on the spool dir, perhaps make it world-writeable and retry as a first test, if
that works you can narrow problems down.
Just my 2 cents
Stefan Märkle
working configuration?
That would be very useful for us 'still-fighting' people ;-)
Stefan
--
Stefan Märkle Netpioneer GmbH
Leitender Systemarchitekt Beiertheimer Allee 18
[EMAIL PROTECTED] 76137 Karlsruhe
to see anything come of it. Just a little note that
Sysmaster is packaging up Asterisk in their product and not
giving a notice with the product with an offer to the source
of the the GPL software they use inside their products. They
even lie and tell people such as myself outright that
From: Matt Riddell [EMAIL PROTECTED]
what's about slashdot?
Matteo
Yup.
1. The strings prove it is Asterisk
2. They deny it is
3. We have multiple sources who can confirm this
So, let's all post the article on GPL violation to Slashdot.
All in favour?
No I'm not.
One of the
But I'm not aware of such a converter.
See my previous mail of the thread (which I quote below)
indicating the URL. The devises are exactly the skype
converter to FXO/FXS. http://www.pcphoneline.com/skype
Sorry for not reading your Mail completely.
So with this thing and an FXO Card in
Sorry, I don't have that kind of skill as well as the time
for development.
So, back to my original question, I'll use the way I proposed
if that is possible.
If skype can be one of the channels, that is an advantage in
holding conference call over the Internet, I think.
Hi,
) lines
plugged to the four BRIs.
Hope someone can point me to a direction where to look further, as we seem stuck right
now.
Stefan Märkle
bri debug span 1
asterisk*CLI bri debug span 1
Enabled
requested on
unconfigured channel 255/255 span 1
Any clues to what happens here?
Seems the communication asterisk=Tenovis does not work. And why is the cause not
handled in chan_zap?
Stefan
--
Stefan Märkle Netpioneer GmbH
Leiter Knowledge Center Beiertheimer Allee
and voicemail functionality.
Just my 2 cents,
Stefan
--
Stefan Märkle Netpioneer GmbH
Leiter Knowledge Center Beiertheimer Allee 18
[EMAIL PROTECTED] 76137 Karlsruhe
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full support by them.
Just my 2 Euro-Cents, and sorry for the long text, hope somebody made it to the end
... ;-)
Stefan Märkle
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Leitender Systemarchitekt Beiertheimer Allee 18
[EMAIL PROTECTED] 76137 Karlsruhe
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