Quyps--
I've noticed in general that the ulaw, alaw, gsm, slin files used and
generated by
asterisk do not have headers (the RIFF stuff), and asterisk is not expecting
them. in general they
will louse up Asterisk's ability to play the sound. They are just raw data
and the extension
on the file
I use Jack for getting callee sound. Dial with option M():
JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on
This works fine, but I need to connect the sound channel to Jack
*before* the actual answer.
As you can see in the AMI log, between Ringing to JACK_HOOK
Does anybody have any experience of running Asterisk with DAHDI on ESXi?
I am running Asterisk 1.4 with DAHDI 2.3 on ESXi 4.0 alongside a TE220
card. My asterisk install can see the card, but no matter what I do
with the jumpers it remains in E1 mode. I have tested the card in
another machine, in
Hi,
How can I convert FROM ALAW file, which generated by asterisk apps
(monitor, or record app), to format (wav or mp3) that is playable by
music player?? Can Sox do this?
I have an asterisk 1.6.2.6 on my CentOS 5.2. I record audio clip by
mixmonitor app and use file command to check the alaw
What does this mean :
[May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644
handle_request_subscribe: Sending fake auth rejection for user
sip:c...@ast_pub_ip;tag=wetpp2qb3f
[May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644
handle_request_subscribe: Sending fake auth rejection for user
It seems to be 401 unauthorized, your end point credentials are not correct
On Thu, May 20, 2010 at 1:30 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
What does this mean :
[May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe:
Sending fake auth rejection for user
The following link may be a suitable workaround
How do I change the type of line from E1 to T1/J1 without using jumpers?
http://kb.digium.com/entry/121/
Alec Davis
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hello.
Can You provide example, how can i run specific extension after incoming
call going into queue and answered (but not hangup).
(i want to use System(echo .) after member of specific queue
answered a call);
Thank You.
--
Vasiliy G Tolstov v.tols...@selfip.ru
Selfip.Ru
--
hi,
i made page for Asterisk T.38 Gateway code testing
http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway
Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming
later BUT Asterisk 1.8 is too far and we need t.38 gw now
if you would like help/test current code(last patch
Which version of asterisk are you running?
Older versions allowed for an AGI to be called when the queued call got
connected with an agent.
core show application queue
Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI])
The optional AGI parameter will setup an AGI script to be
On Mon, May 17, 2010 at 10:26:18PM -0300, Daniel Bareiro wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration that
I'm using in chan_dahdi.conf
This week on VUC:
12 Noon EDT: Office KONNECT - phones that can connect to asterisk or
be used without a pbx
1 PM EDT: Dan York on his new book 7 Deadliest UC Attacks
and the usual segments of VoIP and Asterisk news, and the VUC 1 minute rant.
Info: http://vuc.me
Conference bridges are active
On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question – wouldn’t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel on its screen.
Information
Tzafrir Cohen wrote:
On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question – wouldn’t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel
В Чтв, 20/05/2010 в 05:49 -0700, Jim Dickenson пишет:
Which version of asterisk are you running?
Thank's for answer. One minute before i found answer -
add membermacro to quesues.conf
I'm use asterisk 1.6
--
Vasiliy G Tolstov v.tols...@selfip.ru
Selfip.Ru
--
Sox v14.1.0 doesn't play with alaw, but AFAIK, Asterisk has this function
(this is from 1.4.30, think 1.6X has same functionality)
CLI help file convert
Usage: file convert file_in file_out
Convert from file_in to file_out. If an absolute path is not given, the
default Asterisk sounds
Your receptionist would wait until your back was turned?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP
Sent: Thursday, May 20, 2010 8:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Thu, May 20, 2010 at 09:24:18AM -0400, SIP wrote:
Tzafrir Cohen wrote:
On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question – wouldn’t it be easier to monitor a web interface than a
phone with 100 lights?
Hi, this didn't seem to work. Is there something I am missing?
dialplan add extension 1234,1,NoOp,hello into default
Extension '1234,1,NoOp,hello' added into 'default' context
-- Added extension '1234' priority 1 to default (0x8e8f520)
dialplan add extension 1234,1,NoOp,hello into test
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met other issues I should include include in my checklist ?
Regards
--
On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Tzafrir.
On Thu, May 20, 2010 at 09:58:26 -0300, Tzafrir Cohen wrote:
I'm testing a telephone connected to FXS port of a Sangoma A200 card.
But I'm observing that callerid is not working. The configuration
that I'm using in chan_dahdi.conf is
I am trying to implement a change to our Dialplan that will thwart
tele-spammers that are calling us with blanked out caller ID.
The caller IDs seem to vary between originating callers when they block
caller ID. I've seen the following:
anonymous
So I'm checking for these. However recently
On May 20, 2010, at 12:43 PM, Myles Wakeham wrote:
I am trying to implement a change to our Dialplan that will thwart
tele-spammers that are calling us with blanked out caller ID.
The caller IDs seem to vary between originating callers when they block
caller ID. I've seen the following:
I don't see anything in the SIP trace related to the warning messages.
Would anyone have any further tips?
Thanks for any help!
On Wed, May 19, 2010 at 9:12 PM, David Cunningham
dcunning...@voisonics.com wrote:
What should I expect see if it is the peer asking us to slow down RTP?
Thanks
Does anyone know if you can use softphones on thin clients? I have a
new customer that wants to use Eyebeam (about 10 users) on a thin client
platform. Each user has a little box on their desk that has a USB port,
mic and headphone jacks and monitor.
I am worried about conflicts
On 20 May 2010, at 18:35, Carlos Chavez wrote:
I am worried about conflicts when running 10 softphones on the same
server since they will all try to use por 5060.
And the fact most terminal services servers/clients still don't support audio
input.. only output..
S
--
I am looking for a way to have an agent execute an attended transfer
using the asterisk manager interface (AMI).
I have been trying to use the dual Redirect from svn trunk. The problem
with this function is that the ExtraChannel does not get redirected
properly afaict.
Now, I am looking for
1. GPXE + HTTP
2. Tiny Core Linux
3. Profit...
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, May
Don't some thin clients run on WindowsCE or Linux/rdesktop?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, May 20, 2010 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Thursday, May 20, 2010 1:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Softphones on thin
This is a drawn-out, but efficient way to fix this problem. Create two
programs. Program 1 reads Master.csv (or whatever you use to store your CDR
in). Reads through CDR and creates a blacklist of numbers and ID's. write
blacklist to a text file or database. Program 2 runs from dialplan as
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in
the spirit of your question:
(1) dialplan conversion
(2) loss of functions like Gosub
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
You will find there are an infinite number of bogus CLID's that these
scumbags use to thwart screening. Such things as invalid NPA, invalid
office code are common. Blank is seldom used any more.
Here in the US at least, even with the do not call list ( federal ) and
various state do not call
I've used HP Thin Clients as embedded hosts for Asterisk. The T5700
models that I have are 1 GHz CPUs, more recent models should be able to
run a soft phone without too much trouble. They all have local USB
ports, making USB headsets as good solution.
Another alternative might be to used a soft
Is it possible to use an Asterisk feature code to transfer a call to a
specific extension?
For instance, if you take a call, and the caller wants to go to a
conference, it would be nice to use a feature code for this, rather
than going through a longer transfer sequence.
e.g.:
- You have a
Is the Java soft phone an open source or obtainable? I am just checking
their site and it seems they only provide service??!!
Their java web based client is built neatly. Would like to test that on my
servers.
On Thu, May 20, 2010 at 3:21 PM, mgra...@mstvp.com wrote:
I've used HP Thin Clients
On Thu, 20 May 2010, SIP wrote:
Even IF you could get a keyboard with lights you could individually turn
on and off (never seen one),
http://www.artlebedev.com/everything/optimus/
Bit expensive though...
Gordon
--
_
--
The Asterisk Development Team has announced the final maintenance releases of
Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These
releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The Asterisk releases for 1.6.0.28 and
On Thu, 2010-05-20 at 17:41 +0200, Olivier wrote:
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
I am still running 1.4 because of this bug:
https://issues.asterisk.org/view.php?id=15129
I haven't tried any 1.6 versions recently; looks like some patches
On Thu, May 20, 2010 at 1:49 AM, Pham Quy qu...@vega.com.vn wrote:
Hi,
How can I convert FROM ALAW file, which generated by asterisk apps
(monitor, or record app), to format (wav or mp3) that is playable by
music player?? Can Sox do this?
From alaw to wav, you can use Asterisk's CLI f
David Backeberg wrote:
meetme CLI arguments changed between 1.6.0 and 1.6.2
Don't know where the delta was, and I haven't looked.
I prefer the new syntax, and especially prefer the 'concise' option,
but it might break features people have built in the past.
Specifically,
1.6.0 'meetme' is
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Which bug number have you reported your issue in?
Leif.
--
_
-- Bandwidth and
Danny Nicholas wrote:
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in
the spirit of your question:
(1) dialplan conversion
(2) loss of functions like Gosub
Can you be more specific about what 1) and 2) mean?
Leif.
--
Greg Woods wrote:
I am still running 1.4 because of this bug:
https://issues.asterisk.org/view.php?id=15129
I haven't tried any 1.6 versions recently; looks like some patches have
been checked in since I last tried it, although the bug is not closed.
So I may have to try it again when I
David Cunningham wrote:
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come across is that the peer is congested and
sending RTCP messages asking us to slow the RTP down. Is there any way
we can
On Thu, May 20, 2010 at 7:14 PM, Alec Davis siva...@paradise.net.nz wrote:
The following link may be a suitable workaround
How do I change the type of line from E1 to T1/J1 without using jumpers?
http://kb.digium.com/entry/121/
Alec,
Thank you, thats worked for me.
Although, the 'insmod
On Thu, 20 May 2010, Gordon Henderson wrote:
On Thu, 20 May 2010, SIP wrote:
Even IF you could get a keyboard with lights you could individually turn
on and off (never seen one),
http://www.artlebedev.com/everything/optimus/
Bit expensive though...
Gordon
Heh. A $2400 keyboard.
Not open source, nor free...but certainly available.
--Original Message Text---
From: bruce bruce
Date: Thu, 20 May 2010 15:33:41 -0400
Is the Java soft phone an open source or obtainable? I am just checking
their site and it seems they only provide service??!!
Their java web based client is
This may be totally irrelevant and it may send you down the wrong track, but I
thought I would mention it:
There is a bug which can prevent recent versions of asterisk from creating
proper headers in WAV files.
The bug shows up on Solaris systems but Linux is theoretically not immune to it.
If
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Which bug number have you reported your issue in?
Leif.
I am
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