Displaying 20 results from an estimated 700 matches similar to: "#exec in #include'd file"
2009 Jun 12
1
AmooCon video recordings online
JFYI and slightly off-topic:
All of the videos we recorded at AMOOCON open-source VoIP conference
(Rostock, Germany, May 4-5) are now available on the web site:
http://www.amoocon.com/
All of them are available in different qualities and formats,
including Quicktime 7, versions for the iPhone and iPod and h.264
which IIRC can be played in MPlayer etc.
100 GB in total. :-)
Philipp Kempgen
2009 May 22
1
/etc/asterisk/startup.d
Does anybody think it would make sense for /etc/init.d/asterisk
to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk
did?
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP
2009 Jan 13
2
Zaptel & multiple kernels
Hi,
If I have multiple kernel sources in /usr/src, e.g.
linux-headers-2.6.26-1-686
linux-headers-2.6.26.custom.1
how does the Zaptel Makefile(?) know which one to pick?
Is it a good approach to compile the kernel first and then compile
Zaptel "manually" afterwards?
Or should I rather put zaptel in /usr/src/modules and use
fakeroot make-kpkg ... modules_image
in the kernel
2009 Jun 08
1
OT: Grandstream, call pickup, ...
Maybe it's just me, but I get the impression that Grandstream is
quite uncooperative.
We (and others) have asked them multiple times to make the call-
pickup code ("**") configurable but either they don't understand
the request or they're unwilling to do anything about it.
http://forums.grandstream.com/node/2848
http://forums.grandstream.com/node/709
Unfortunately their
2009 Feb 21
0
Where to find db1_dump185 in debian packages ? [SOLVED]
2009/1/30 Philipp Kempgen <philipp.kempgen at amooma.de>
> Olivier schrieb:
> > Here http://www.voip-info.org/wiki/view/Asterisk+database , you can
> read:
> > "Also, since it's a normal Berkely db1 (version185) file its contents can
> be
> > viewed/dumped with the standard *db1_dump185* tool. Thus db1_dump185 -p
> > /var/lib/asterisk/astdb will
2009 Jun 16
2
Update Caller-ID after Dial()
Can you confirm that currently there is no way to update the caller
ID via the manager interface once the B leg is ringing or connected?
Looks like this would be feasible with the functions introduced in
https://issues.asterisk.org/view.php?id=8824 ("[patch] Remote (called)
Party Identification - chan_sip & chan_skinny implementation").
Such functionality could be desirable in
2009 Feb 14
1
Progress() and Proceeding()
Hi,
The descriptions of Progress() and Proceeding() are really vague.
Progress():
---cut----------------
[Synopsis]
Indicate progress
[Description]
Progress(): This application will request that in-band progress information
be provided to the calling channel.
---cut----------------
Proceeding():
---cut----------------
[Synopsis]
Indicate proceeding
[Description]
Proceeding(): This
2009 Feb 19
1
queue_variables() function
Hi,
Can somebody please shed some light on how to use the
QUEUE_VARIABLES() function?
The built-in help says
---cut---
Return Queue information in variables
[Description]
Makes the following queue variables available.
QUEUEMAX maxmimum number of calls allowed
QUEUESTRATEGY the strategy of the queue
QUEUECALLS number of calls currently in the queue
QUEUEHOLDTIME current average hold time
2009 Apr 27
0
SIP infrastructure
O boy. SIP infrastructure is so flexible that basically nobody gets
it right. :-)
You could easily have 20 different SIP network elements (/servers
/services). Even more.
And we get at least 5 new SIP-RFCs per day. They're all trying to
fix things which the previous specifications didn't address. :-)
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany ->
2009 May 20
0
dtmf=info and canreinvite=yes
Hi,
Sorry for this "newb" question (but maybe a newb question once in
a while is ok):
What's the current state about Asterisk handling DTMF features via
SIP INFO (dtmfmode=info) even when the media path has been reinvited
(canreinvite=yes) to go directly from one phone to another?
Somewhat related to this suspended issue:
https://issues.asterisk.org/view.php?id=14126
How widely
2009 Nov 29
3
Parsing custom SIP headers
Hi,
Just to be sure: Is there a dialplan function in Asterisk that
parses custom "name-addr"-style SIP headers for me?
If I wanted to do it right the syntax
name-addr *(SEMI generic-param)
is quite complex to parse in the dialplan using nothing but CUT().
It's so easy to make false assumtions about angle brackets (< >),
whitespace (LWS), quotes (") around the
2009 Jul 26
2
Verbose() messages go unnoticed
Does anybody else have the feeling that custom messages
(Verbose(1,...)) do not stand out enough on the CLI?
We're sending messages like "Extension 123 is unknown" to the output
and that should tell the user why a call to 123 fails but users fre-
quently crank up the verbosity to 3 or 10 so our messages go unnoticed
in many cases.
My idea was to use terminal escape sequences to make
2009 Jun 03
0
RES: RES: SIP Response 181 - Is it possible in A steri sk?
Hello Philipp and All,
My scenario is a bit different than the one I had explained before. I'm
sorry.
Let's suppose I have someone calling one of my Asterisk clients. This
asterisk client has CFB (Call Forward Busy) activated. The forward number is
a Voice Mail System, however is not a Asterisk's Voice Mail.
It is a third party Voice Mail System, that has a SIP Trunk with my
2008 Dec 02
0
Using Dial M option from extensions.ael [SOLVED]
2008/12/2 Philipp Kempgen <philipp.kempgen at amooma.de>
> Philipp Kempgen schrieb:
> > Olivier schrieb:
> >
> >> How can you use Dial application M(x) option from extensions.ael ?
> >> (As a reminder, this M(x) executes macro x when Dial called party
> answers).
> >>
> >> It seems to me that asterisk keeps looking for this macro in
>
2008 Dec 18
0
Latest AstManProxy [SOLVED]
2008/12/18 Philipp Kempgen <philipp.kempgen at amooma.de>
> Olivier schrieb:
>
> > I unsuccessfully tried to download AstManProxy, clicking over download
> > button in http://github.com/davetroy/astmanproxy/tree/master .
> > It failed with "XML error".
>
> Try again. It works.
You're right : now it works !
I can't explain why it didn't
2008 Oct 27
0
make config update-rc.d on Debian
This was an old thread
http://lists.digium.com/pipermail/asterisk-users/2007-November/200539.html
so I'm starting a new one.
Tzafrir Cohen wrote:
> On Thu, Nov 15, 2007 at 06:47:04PM +0100, Philipp Kempgen wrote:
>> On Debian the Asterisk Makefile does
>>
>> /usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .;
>>
>> which results in a
2008 Nov 29
0
pp_each_user(), pp_each_extension()
---cut-------------------------------------------------
-= Info about function 'PP_EACH_USER' =-
[Syntax]
PP_EACH_USER(<string>|<exclude_mac>)
[Synopsis]
Generate a string for each phoneprov user
[Description]
Pass in a string, with phoneprov variables you want substituted in the format of
%%%%{VARNAME}, and you will get the string rendered for each user in phoneprov
2008 Aug 17
0
asterisk -n switch
man asterisk (1.6.0-beta9) says:
> -n Disable ANSI colors even on terminals capable of displaying them.
But on the CLI (asterisk -n -r) any "core show application"
or "core show function" leaks color escape sequences.
Gr??e,
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied ->
2008 Sep 15
0
[OT] email netiquette (was: Re: Re: Asterisk realtime MySQL clients from same IP problem)
Your right with this part
But as I also have some knowldge on other parts but ms , *nix etc I know it is nowadays possible for almost every email client to correctly display html email. And be honest does it not read more easy if you have a nice font and some markup available?
I know mailman is an old package and should be more flexible in handling and distributing html email.
For standards:
2008 Nov 23
1
SendImage()
SendImage() in 1.4:
---cut---
SendImage(filename): Sends an image on a channel.
If the channel supports image transport but the image send
fails, the channel will be hung up. Otherwise, the dialplan
continues execution.
The option string may contain the following character:
'j' -- jump to priority n+101 if the channel doesn't support image transport
This application sets the