Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions ?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I
2009/5/11 Olivier oza-4...@myamail.com
2. Secondly, I would like not to use GOSUB_RETVAL and call a custom
function just like I'm calling other functions with statements like :
123 = {
NoOp(TOLOWER(fOo BaR));
Here, I meant
NoOp(read this ${TOLOWER(fOo BaR)});
Hello,
It seems /* */ comments are not supported in ael.vim (which brings AEL
syntax-highlighting to vim).
Is it hard to add this feature and have uploaded in vim extensions
downloading site ?
Regards
___
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John F. Ervin wrote:
| So, people have recommended building a system from scratch, start with a
| CentOS base and installing asterisk and all of the other utilities.
| I've only used Trixbox for my business system. I'm wondering what
| surprises I'd
On 11/05/09 04:21, John F. Ervin wrote:
snip /
Are there (??) instructions for people who are experienced at the
Trixbox level but wish to move on?
Sure, the TFOT book is a great start. If you want to use Ubuntu or
Debian rather than Centos then Asterisk is in the Debian and Ubuntu
Server
Olivier schrieb:
It seems /* */ comments are not supported in ael.vim (which brings AEL
syntax-highlighting to vim).
Are C-style comments supported in AEL? I don't think so.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
On Mon, May 11, 2009 at 1:55 PM, Philipp Kempgen
philipp.kemp...@amooma.de wrote:
Olivier schrieb:
It seems /* */ comments are not supported in ael.vim (which brings AEL
syntax-highlighting to vim).
Are C-style comments supported in AEL? I don't think so.
They are.
Regards,
Atis
--
Atis
2009/5/11 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
It seems /* */ comments are not supported in ael.vim (which brings AEL
syntax-highlighting to vim).
Are C-style comments supported in AEL? I don't think so.
This page says it does
Hello,
I have so far Polycom 501 and 403 which displays the label of a key and
the name of a buddy near its key. Today I received new 330's and they do not
display the name, only 1 2. Besides that they work correctly. Anyone
has an idea, or is it a known feature?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi,
I am using Asterisk 1.6.0.9. I have calls coming from another asterisk
server via IAX and lands in a queue. I have noticed that if there are
no agents logged in the queue no CDR is generated. If there is one
agent logged in then the phone rings and a CDR is generated even if
the call was
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the
This is the most useful script anyone has published on this list for a long
time.
Thanks David, * stars on this..
I can finally have our clients move from trixbox to an asterisk vanilla
system in no time now.
Ps.. here are a few suggestions..
àMove ntpdate ntp.bri.connect.com.au out
2009/5/11 Watkins, Bradley bradley.watk...@compuware.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing
The Asterisk Development Team is pleased to announce the second beta of
Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at
http://downloads.digium.com/pub/asterisk/
This release merges in changes to the device state code which caused a
performance regression in
On Mon, May 11, 2009 at 02:16:29PM +1000, Klaverstyn, David C wrote:
Hi John,
I'm not sure if this will help you or not but I created a script that
will install Asterisk with all the required components for DAHDI,
Faxing, fax to email, LDAPget, CDR, FOP etc. It can even include text
out there is a file to change the dtmf duration
where are you? or from where is your cellphone?
from other phones like lkand lines it works well?
David
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile
Message: 10
Date: Fri, 8 May 2009 20:30:11 -0700
From: Eric Fort eric.f...@gmail.com
Subject: [asterisk-users] VoIP over satellite internet
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
On Mon, May 11, 2009 at 1:30 AM, Olivier oza-4...@myamail.com wrote:
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions ?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
snip
...routing via satellite adds about a quarter second of latency to the path.
Is that too much?
/snip
Eric,
I believe that you are mistaken. Routing via satellite adds about a quarter
second of latency PER TRIP from earth to orbit. This is simply due to the
distance a satellite is from
David Gibbons wrote:
snip
...routing via satellite adds about a quarter second of latency to the path.
Is that too much?
/snip
Eric,
I believe that you are mistaken. Routing via satellite adds about a quarter
second of latency PER TRIP from earth to orbit. This is simply due to the
On Fri, May 08, 2009 at 11:56:42PM -0500, Frank Bulk wrote:
If people don't mind taking turns talking, it will work. It's just going
to be like talking on a CB. Reminds me of talking to my grandparents in the
Europe as a child in the early 80's.
Just recall that VoIP can generally live
Anyone using Nokia E Series handsets with Asterisk? I'm trying to
deploy some e71's and am having an issue. I can get a single handset
working, but when I try to create a SIP profile on the second phone, it
won't allow me to save the profile, saying that devices in the same
realm must have
snipOf course, that's assuming your satellite is in geosynchronous orbit. If
It's in LEO.../snip
Singer,
You are of course correct, low earth orbit will have lower latency. I was
assuming that this user would be using a stationary link on the ground, not a
portable sat phone or an aimable
Cory Andrews a écrit :
Anyone using Nokia E Series handsets with Asterisk? I'm trying to
deploy some e71's and am having an issue. I can get a single handset
working, but when I try to create a SIP profile on the second phone, it
won't allow me to save the profile, saying that devices in the
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as dtmfmode=auto. Calling from mobile phone or
Good morning,
I'm having suddenly cut-offs and I don`t know why. It's been hapenning since
I enabled cdr_odbc/func_odbc in my system.
We use func_odbc to register some queue member's events (login, pause, etc.)
at an external DB ('remoto' connector) and to uptade local tables at a local
DB
Hi Cory,
I believe you meant that you can't add a second SIP profile on the same
phone, right? There seems to be a bug with the latest Nokia E-series
that has this problem. It complains even if the userid and password are
identical.
I worked around this by just changing the realm name. Nokia
Hi All,
I'm at the end of my tether here and would really appreciate some help.
I'm trying to implement DTMF based pause/resume of call recording. I'm
using Asterisk 1.4.22.1.
Here's the scenario:
The caller (SIP or ISDN, doesn't matter) dials into the asterisk which
executes the
Jon Morgan wrote:
Hi All,
I’m at the end of my tether here and would really appreciate some help.
I’m trying to implement DTMF based pause/resume of call recording. I’m
using Asterisk 1.4.22.1.
Here’s the scenario:
The caller (SIP or ISDN, doesn’t matter) dials
I loaded PBX in a flash and I have a simple dialplan setup. I'm guessing this needs to go on the DMZ of my router for anyone to get to it correct? Is there any way to keep it behind the router and map to it or is that more trouble than it is worth?Thanks!Ronny
On Monday 11 May 2009 19.54.47 k4...@bellsouth.net wrote:
I loaded PBX in a flash and I have a simple dialplan setup. I'm guessing
this needs to go on the DMZ of my router for anyone to get to it correct?
Is there any way to keep it behind the router and map to it or is that more
trouble
For my information (and anyone else interested), how much of the information
at this link - http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf
is still valid?
According to that information, the setup you describe would basically allow
for 250 or so concurrent calls. Also, I expect that
Other SIP clients behind the firewall (not using STUN, work).
We have a SIP client using STUN and ICE behind a pfSense firewall.
The firewall is behaving oddly.
REGISTER packets work fine.
But when the client tries to make a call, the first INVITE packet from
the client pass through the
On Fri, 2009-05-08 at 20:30 -0700, Eric Fort wrote:
Could those on the list who have used or tried to use VoIP over a
satellite internet connection comment on how well it works or if it
even works at all in a reliable way. What is the effect of latency on
the VoIP path and how much is
- Eric Chamberlain e...@rf.com wrote:
Other SIP clients behind the firewall (not using STUN, work).
We have a SIP client using STUN and ICE behind a pfSense firewall.
The firewall is behaving oddly.
REGISTER packets work fine.
But when the client tries to make a call, the first
On Monday 11 May 2009 10:54:48 am Daniel - Asterisk wrote:
*Realtime peers are reacheble again, why they got unreachable?*
[May 11 09:02:59] NOTICE[17835] chan_sip.c: Peer '870' is now Reachable.
(27ms / 2000ms)
Unless you saw a message that said the peers became unreachable, then they
were
Administrator TOOTAI wrote:
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as
On May 11, 2009, at 2:30 PM, Tim Nelson wrote:
pfSense employs source-port randomization by default. You may want
to enable advanced outbound NAT which turns this behavior off.
While I'm not sure this is the source of your problems, I've seen it
ruin otherwise acceptable SIP situations.
For those also need NT over PtMP, i started a initial patch for it. Very
limited at the moment, only one incoming call to chan_dahdi from one
device is possible. But i was pleasantly surprised that NT-ptmp is working
anyway
Get the patch here: http://bugs.digium.com/view.php?id=15048
Kristijan
I got very excited when I read the title of this email - I was hoping
someone had learnt to speak g729.
Ah well.
PaulH
Adrian Marsh wrote:
Hi,
I’m having problems with an asterisk server that’s not offering Codecs
for ulaw and alaw as it should.
I’ve three servers in total: a1, a2 and
All,
I think we've found what was blocking us. It seems that SElinux, for
some unknown reason, didn't like the AMR codec, and did something to
block it.
Set that to passive, and the problem goes away...
Would still like to learn more about asterisk codec translation though,
if anyone has
Has anyone else had issues with Originate returning the wrong error code?
According to the docs, the following errors are supposed to be returned:
0 = no such extension or number
1 = no answer
4 = answered
8 = congested or not available
Now in Asterisk 1.4.23 I get some error code 5's but since
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