recording problems with Asterisk
Hyper-V VM. Are problems actual?
Thank you!
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Dmitriy Ermakov.
On 24 July 2016 at 18:40, Dima Ermakov wrote:
> Good day!
> Could you help me with my question?
> I want to install AsteriskNow as Virtual Machine on Hyper-V (Windows
> Server 2012).
&
> On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote:
> > Setting the user's shell to /usr/sbin/rasterisk works. On login user
> > gets into asterisk CLI if asterisk is running (user just has to have
> > write permission to /var/lib/asterisk.*).
>
> How does th
Setting the user's shell to /usr/sbin/rasterisk works. On login user
gets into asterisk CLI if asterisk is running (user just has to have
write permission to /var/lib/asterisk.*).
> On Fri, Oct 31, 2008 at 11:11:08AM +0100, fadey wrote:
> > Hi, everyone
> >
> > I'm investigating if I could give a
Hello, everyone
I'm looking for a way to find out if there is a device registered on a
particular extension from dialplan. Does anyone know how to do that?
Thanks in advance.
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Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes, when making a call, my asterisk box doesn't detect a ringback
and I just hear silence until the other party picks up the phone. I've
checked the SIP messages and they are ok (I'm getting 183 "session
> My specific problem is that Asterisk 1.2.17 and 1.2.18 (I've not tried
> 1.2.16) core dumps at least once per day. 1.2.15 works just fine for
> me. I don't know if there are open bugs. I've not opened any bugs.
> Any time I open a bug for a problem I have on a production server, all
> peop
> I'm running a
> 1.2.7.something (i think) that has been running almost nonstop since
> installing. Very reliable and stable for my needs.
This version has some security issues inside.
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aster
ay 2007 09:44, dima wrote:
>> Hello, everyone.
>> I'm having a strange problem with my asterisk. After dialing and
before the other side picks up the phone I should hear the tones (I'm
not sure what are they called: p---pii) and
> > in almost al
Hello, everyone.
I'm having a strange problem with my asterisk. After dialing and before
the other side picks up the phone I should hear the tones (I'm not sure
what are they called: p---pii) and in almost
all cases that is true. However there is a range of numbers where
Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every
time I reload asterisk I get this in CLI:
-- Reloading module 'app_playback.so' (Sound File Playback Application)
[May 3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading
say.conf
== Parsing '/etc/asterisk/
> Do you have the context numberplan-custom-1 in your extensions.conf file? I
> think if you don't have it in extensions.conf then it goes back to using
> default.
Yes, it is defined in extensions.conf
extensions.conf
.
[default]
exten = _X.,1,NoOp("This is default")
[numberplan-custom-1]
3b0' status is
'UNKNOWN'
Another thing I didn't mention is that I used GUI for the initial
configuration.
>
> dima wrote:
> > Tnaks for your answer. Sorry, if I'm missing something obvious here.
> > Yes, it's asterisk 1.4. I've configured a
: sip:[EMAIL PROTECTED]:5060
> Is this Asterisk 1.4.x?
>
> from samples/extensions.conf...
> ;
> ; User context is where entries from users.conf are registered. The
> ; default value is 'default'
> ;
> ;userscontext=default
> ;
>
> Is this any help?
>
Hello, everyone.
Today I've installed an asterisk svn trunk (r61667). The problem I'm
having is no matter what context I set in the config file for that peer,
"default" is always being used.
The output of "sip show peers" shows the context correctly, but when I
try to make a call, using that peer,
Indeed, the IRQ priority of my card is low. Sorry for being lame, but
does that have to be set in BIOS? Are there any "best practices" for
choosing an IRQ or I can set it to any free number?
> I would start with IRQ sharing. Make sure that your Zap hardware isn't
> sharing an IRQ. Secondly, you
Hello, everyone.
I get
Best: 99.975586 -- Worst: 99.975586 -- Average: 99.975586
with zttest
Where do I have to start looking for hardware errors?
Thanks in advance
> younss azzayani wrote:
>
> > and this is the /var/spool/hylafax/log/c1:
> > http://pastebin.ca/403282
> > cat /var/spo
perhaps you should try
pear install DB
However note that this mailing list has nothing to do with pear.
> Hi, i try install FreePbx by tuturial in
> http://www.voip-info.org/wiki/view/Instalaci%C3%B3n+de+Asterisk+en+Ubuntu+Server+Paso+a+Paso&view_comment_id=13443
>
> but i have this error when i
Hello, everyone.
I'm having a small problem when using asterisk with GUI. For every
provider I create I have to set "insecure=invite,port" in users.conf. Is
there a way to make it a default setting?
Thanks in advance.
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Hello, everyone.
I'd like to ask how does asterisk 1.4 with GUI register itself at the
provider's end (when I mark a checkbox 'register' while creating a
"Service Provider"). Before I used to write something like:
register => 924980111:[EMAIL PROTECTED]/924980111
in sip.conf. Having that line, aste
Hello, everyone.
I'm reading about the asterisk new features. One is T.38 protocol
support. I used faxes before with asterisk 1.2 and everything was
working quite well. Could anyone explain what have changed in the way
faxes are handled.
Another thing is, in order for asterisk to work over T.38 wit
Hello, is there any way to reduce voicemailmain functionality without
recompiling .c file? Is there any external .conf file i can use to do it?
For example: i want to restrict password changing for users, etc.
Thank you for attention.
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Hi all:)
Can you answer,how to change parameters of VoiceMailMain application?(for
example:i dont want to give permission to change password and etc)
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i have problem with dial-plan in php. I have 3 extention in dial plan, 555,551
and 551. the problem is that READ PIN||3 works in 555 and hangs in other
extentions. (after timeout asterisk writes thar "user entered nothing"). i
can't get what's wrong...
here is my dial-plan
[incoming]
exten =>
How to limit the duration of the MeetMe conversation?
--
thank you
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How can I configure asterisk to switch from my hardphone which is always
up and online, as soon as I register with my notebook's softphone on the
asterisk server?
The target is to receive all calls destinated to my hardphone on my
softphone when it's online.
Any ideas?
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