of the page declared: 1 minute is defined as 57 seconds.
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).
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...
Have a nice day,
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can get this chan-spy application
Maybe it's been replaced by the Monitor app?
Or does it do something else?
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. Maybe you have me confused with somebody else...
There is still a problem somewhere with somebodies mail gateway where
one can receive multiple copies of the same email. Your email reached
me three times for example.
Again, my apologies,
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+authentication
It should tell you all you need to know...
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on the spool
directory and can read all the parent directories...
Hope this helps,
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of the components has been replaced by a file or directory
with bad permissions
Hope this helps,
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out of the
box, first time, no hassles. How can you compete with that.
Asterisk is a PBX, Skype is not, but Skype solves a problem simply and
straight forwardly that Asterisk doesn't.
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to play MOH.
I don't do anything with music-on-hold, but...
[moh_files]
default = /var/lib/asterisk/moh-native
^^^
That spacing looks really odd...
Hope this helps,
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quite understand the benefits, but this is the way
things are.
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, and the other files (eg. sounds) can be owned by root and
made world-readable.
The only issue is that as non-root, asterisk can't set the TOS bits
(though iptables can do it for you) and it can't set its own priority
(though nice/renice can help there).
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one that worked. Not just that, it worked
*perfectly* first time, no whacky obscure problems.
Now if only Firefly worked under linux, that's be really cool...
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with mailing lists is longer than the list
of projects with forums...
Feel free to set one up, but if the mailing list is removed for a
forum, many current people won't subscribe...
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) and you don't have to
download all the messages...
Whether this list can work like that I don't know, but there it
certainly works.
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.
Have a nice day,
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. Time being
money completes the picture.
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way to preserve the registration
across a reinstall? Perhaps by backing up a directory or a file? Any
help appreciated.
Check the archives to be sure, but I beleive the relevent files are
stored in /var/lib/asterisk/licences or something like that.
Good luck.
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...
Dial(SIP/904)calls whoever logged on as john.
Any ideas?
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A google search shows exactly one reference, so it appears to exist
somewhere. It's in somebodies CVS, any ideas?
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failing
until it finds something that does exist, the default context, which
hangs up on you.
You need to set the context for incoming calls. Look for distinctive
rings.
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DCall: 0 [203.194.19.82:4569]
glenmorangie*CLI iax2 debug
IAX2 Debugging Enabled
At this point the other machine starts recording this one as
UNREACHABLE. I've disabled all trunking, but that didn't help.
Thanks in advance,
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ports SIP is going
to use...
I have tried and tried and tried to get this working but with no luck
(well, I can get it to work with canreinvite=no, but thats not what I
want. I want * out of the audio path)
Good luck!
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examples of guest users,
that don't need to login. No username, no password. Generally dropped
to an incoming only context. After all, the idea is that anyone should
be able to call you without having an account on your server.
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, the email seems to imply that the TE405P will be fine,
though it doesn't say that explicitly.
Basically, is anyone using a 4 port E1 card successfully on an IntelĀ®
E7221 Chipset or similar?
Thanks in advance,
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also tried SJ-Phone and it happens with that as well.
I don't know if you're using linux or not or exactly how the code is
structured, but... if send() or write() to a network socket returns
-EPERM it's generally that your firewall blocked it.
Hope this helps,
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it.
It's a harmless message...
I'm fairly sure that IO-APIC only systems don't have this problem.
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On Fri, Feb 11, 2005 at 06:09:06PM +0900, ?? wrote:
If you actually sent text instead of an HTML only email, you increase
the chance someone will actually read your message...
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a
similar effect by unloading and then reloading the module. In my case I
was testing H323, it might be trickier if you're actually using what
you're playing with...
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the new jitter buffer makes it in too..
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at all. So you always start in s.
Hope this helps,
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they
require completely different versions of OpenH323.
Thanks in advance,
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about how to do it.
I was thinking about letting people provide a quality rating, maybe 1-5
on call quality. This would allow people to compare price/quality and
aim for where they feel comfortable.
But I think it's an awesome idea...
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will be appreciated.
I am using libpri verision 1.0.4 with Fedora Core version 2.6.5-1.358.
Note: Asterisk and the kernel modules compiled fine, I would just like
to try out this utility.
Thanks,
-Arlen Raasch
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error you're getting...
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/pipermail/asterisk-users/2004-September/062872.html
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that the hardware doesn't require changing, this is
good...
Thanks in advance,
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of it
- The actual binary shared library is modified
- The system contacts Digium every time you start asterisk
In the last case nothing is changed at all and I'm fine.
Thanks in advance,
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.
So what will???
Thanks in advance,
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for a particular call?
In any case, if it can't be done this way, could you explain why?
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faxes.
Faxing across the world does work so there is something else going
here...
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On Fri, Feb 25, 2005 at 09:15:19PM +0100, Martijn van Oosterhout wrote:
You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from
/lib/asterisk which I have as a real disk...
Even if you can reregister each time, it easier to remember the actual
licence than it is to remember the key and reenter manually each time.
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to diagnose your
problem...
Have a nice day,
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thinking
of implementing it though?
Have a nice day,
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with the problem.
Have a nice day,
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might do a better job...
That aside, if they'd acquired licences for codecs or things, it might
be tricky. Then again, if you paid for the licence when you bought the
phone why would they care.
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...
Then they can use 6000 and 6001 to call themselves and eachother.
This should be enough to get you started.
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=4,1,0,ccs,hdb3,crc4
Hope this helps,
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to make a list. There are lists around
the place, telling you what each prefix means. It also changes over
time, as countries change their numbering to deal with growth of
population and services.
http://www.wtng.info has a lot of useful information...
Hope this helps,
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codes are three digits long. Some are two.
e.g. UK 44 , Bermuda 441
Does anyone know a formula for determining which part of a dialled number is
the country code and city code ?
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decided to search.
Hope this helps,
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,
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On Mon, Mar 07, 2005 at 12:10:48AM +, Mike Dent wrote:
BT providing IAX2 and SIP termination? Hmmm, maybe one day.
Telstra (BTs equiv in Australia) is trialling a VoIP service.
Unfortunatly, it's not quite clear what services they'll be
providing...
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accedently start it as root.
Doing it once by accident tends to alter permissions on files that
then needs to be fixed manually...
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sufficient of these errors, decides the
connection is dead and notifies asterisk.
Although, with UDP (in Linux anyway) the error can be passed back.
Strange problem though, how can only some packets generate Port
Unreachable, but not all. Random routing problem?
Have a nice day,
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headers on the system
as the exact kernel headers I compiled the kernel and I'm running that
kernel also. Has always worked for me...
Hope this helps
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,
like the alsa channel, or a zap channel. But I'm just thinking of if
they've selected an option that allows them to dial a normal number, to
also provide a normal dialtone. Should I just record one and use
Background()?
Thanks in advance,
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a failure later.
FWIW, you probably wanted IAX2/target/3479450772.
Hope this helps,
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++ 3.0.4,
gcc 2.95.4)?
Are there any clients (other than ohphone) which one can use with asterisk
to test it out?
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Is there a way to register with or call the firefly network from an Asterisk
server. It would be pretty cool if you could gateway calls onto it.
Have a nice day,
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an issue with openh323 where it didn't use the right c++
compiler, but that's just my setup.
Hope this helps,
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the codec converters, or set their cost very
high, would that help?
Given that the cost of no conversion is by definition zero, it's a bit odd
that asterisk is changing them at all unless absolutly required. There must
be something else going on...
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command is an upgrade?or Is there some problem in
my installation?
Dial command is provided by the OSS and/or ALSA modules. Check you're
loading one of those...
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