On Tue, Oct 7, 2008 at 2:20 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM
On Tue, Oct 7, 2008 at 8:45 AM, Pavel Jezek [EMAIL PROTECTED] wrote:
Steve Murphy wrote:
On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
Atis Lezdins wrote:
On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
Hi, according to discussion on asterisk IRC, where
had passtrough mode and 1.6 can send and receive.
Regards,
Atis
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Regards,
Atis
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it every day.
On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.
Hopefully it works. The one in CallWeaver doesn't.
How do you mean - it doesn't? We currently use CallWeaver - Asterisk
1.4 - SIP Provider
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app_queue.so would do the trick :)
Regards,
Atis
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.
Regards,
Atis
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On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is completely illegal in any country
On Wed, Oct 1, 2008 at 5:09 PM, Steve Underwood [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Sun, Mar 23, 2008 at 11
0
7 modules loaded
And add in modules.conf:
noload = cdr_csv.so
noload = cdr_odbc.so
noload = cdr_pgsql.so
noload = cdr_sqlite.so
noload = cdr_sqlite3_custom.so
for each module not used.
Regards,
Atis
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, recompile Asterisk with DONT_OPTIMIZE and then load
core file in gdb and launch bt full.
For more info see doc/backtrace.txt in asterisk source directory.
You can search for existing problems in bugs.digium.com or post here if unsure.
Regards,
Atis
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there was some weird application level support.
Does chan_mobile supports video too? Would it be possible to have 3G
adapter and interact with it?
This just brings Asterisk to new level :)
Regards,
Atis
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[EMAIL PROTECTED]
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Cell Phone
need to continue the execution if the
caller hangs up first too.
What do I need to do?
Search for h extension
Regards,
Atis
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before you dial to
destination peer. For example:
Monitor(ulaw,/tmp/recording-${UNIQUEID},b);
Regards,
Atis
Kindly give your suggestion on this.
Asterisk version - 1.4.21.2
Thanks,
balasam.
On Tue, 09 Sep 2008 Atis Lezdins wrote :
On Tue, Sep 9, 2008 at 3:19 PM, bala krishnan [EMAIL
On Tue, Sep 9, 2008 at 1:22 PM, Thomas Winter [EMAIL PROTECTED] wrote:
On Monday 08 September 2008 14:44, Atis Lezdins wrote:
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED]
wrote:
I dont have problem to make a reload by AMI.
My questions was if module reload app_queue.so
Atis Lezdins wrote :
On Mon, Sep 8, 2008 at 11:39 AM, bala krishnan [EMAIL PROTECTED]
wrote:
Hi,
To disallow the native bridge between the zap channels, i enabled the t
flag in the Dial application. But i dont want to allow the callee/caller
to
transfer the call.
Why would you need
On Mon, Sep 8, 2008 at 8:37 AM, Thomas Winter [EMAIL PROTECTED] wrote:
On Sunday 07 September 2008 21:49, Atis Lezdins wrote:
On Sun, Sep 7, 2008 at 4:56 PM, Thomas Winter [EMAIL PROTECTED]
wrote:
is not work for periodic-announce-frequency and periodic-announce.
An reload is necessary
media processing away
from your CPU.
Alternatively you can enable Monitor/MixMonitor, it should keep
Asterisk in media path.
Regards,
Atis
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Cell Phone: +371 28806004
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Work phone: +1
reload.
As for executing CLI commands, see manager action Command:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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| MEMBERALREADY | NOSUCHQUEUE
Example: AddQueueMember(techsupport|SIP/3000)
Regards,
Atis
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Cell Phone: +371 28806004
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it to any value, so that device state events are
generated, so set it to 10 or 20 to have no actual limit.
Regards,
Atis
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[EMAIL PROTECTED]
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Cell Phone: +371 28806004
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from web server to Asterisk server.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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deadlock problem in
state_interface, however that shouldn't keep you away, as we have 2000
calls per day and we've seen it only once for half year. I hope it
will be fixed soon.. (putnopvut?)
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
On Tue, Aug 26, 2008 at 7:26 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
Are there any plans to back port this feature into upcoming 1.4
releases?
No, new features are added only in trunk, and released in next major
release (1.6).
So
handshake whenever
they detect fax on line. Looking into specs, says me that 2801
supports T.38, so perhaps it could be better idea (altough you would
have to use Asterisk 1.6 and app_txfax for sending faxes)
Also Hylafax log could say something.
Regards,
Atis
--
Atis Lezdins,
VoIP Project
be fairly simple. Also i would suggest subscribing to
asterisk-svn and watch for commits to app_queue to not miss any
bugfixes to it.
Migration to 1.6 could be more time consuming, as there are lot of
changes, you will probably have to adjust dialplan, etc.
Regards,
Atis
--
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VoIP
On Tue, Aug 26, 2008 at 5:39 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote:
I'd say - go for backport instead. shared_lastcall is commited in
http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985
and it seems
[01],2,Return()
; and so on, just better reorganize your extensions so that this can
match patterns better.
[dial-out]
exten = _9.,1,GoSub(clid-mangle,${CALLERID(num)},1)
exten = _9.,2,Dial(SIP/provider)
Regards,
Atis
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Cell Phone: +371 28806004
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Work phone: +1 800 7502835
download it here
http://ftp.iq-labs.net/pbx-test/
If you find it useful, or get into some problems, don't hesitate to write me.
If you need just bunch of identical calls, you may also try out SIPp.
Regards,
Atis
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[EMAIL PROTECTED]
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of call. A little clutter, but it works more or less.
Regards,
Atis
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Cell Phone: +371 28806004
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t38pt_udptl=yes
[callweaver]
type=friend
host=127.0.0.1
permit=127.0.0.1
context=callweaver_out
port=7060
allow=all
canreinvite=no
t38pt_udptl=yes
; note - SIP provider don't have entry, it's dialed by IP.
Regards,
Atis
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[EMAIL PROTECTED]
Skype
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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unsuccessful on that part.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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=x
context=fax
permit=127.0.0.1
allow=all
P.S. after editing inittab, you also have to execute
# kill -HUP 1
So that init process re-reads configuration.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
://lists.digium.com/mailman/listinfo/asterisk-users
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everything that's available from web.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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Now: http://www.astricon.net
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Cell Phone: +1 800
to cell phones by cable, however it supports also skype
(just 1 account). It will launch fake X server and original skype, and
communicate with it.
http://www.celliax.org/
Regards,
Atis
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[EMAIL PROTECTED]
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Cell Phone: +371
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somebody with penalty 2. Then, if dialed member(s)
don't answer, queue will again try somebody with penalty 1 first.
Regards,
Atis
Thanks
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis
Lezdins
Sent: Monday, August 04, 2008 2:29
explaining how to do this by adding custom code to
Asterisk sources, and I guess it could be already done in trunk.
Regards,
Atis
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Cell Phone: +371 28806004
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. :)
There is QUEUE_MEMBER_COUNT (in 1.4) and QUEUE_MEMBER (in 1.6)
dialplan functions which allows to get count of members (in 1.6 also
count of free / logged in members). You can use GetVar to evaluate
that.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
here.
Regards,
Atis
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show just
duration and billsec (at least for 1.4), so i would defineately want
this 1 second between 3 and 4 to show up in some record (preferrably
in second CDR, as it's not talking time with first user anymore).
Regards,
Atis
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[EMAIL PROTECTED
with that? This fits perfectly for my needs. Is there a
way how to exploit this?
Regards,
Atis
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state_interface device when logging in agents.
For more information please search for asterisk queue state, as this
has been discussed several times.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1
bug reporting, improvement suggestions, hell I debug and
report on the entire new CDR/CEL branch :)
ROFLno seriouslyI want one ;-)
How about sending those out when certain amount of karma is reached? ;-)
Regards,
Atis
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[EMAIL
On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Matt Florell wrote:
Hello,
I guess I am one of the lucky few to have one
duration). Mix
and transcode (to some lower bandwidth codec) the rest of recordings
at night time.
Personally I record everything in ulaw, and either on listen or at
night transcode to gsm for storage.
Regards,
Atis
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[EMAIL PROTECTED]
Skype
.
Regards,
Atis
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/sh
/usr/sbin/asterisk -rx 'logger reload' /dev/null 21
logger reload rotates logs. But not CSV . That's because the CSV CDR
files are not held open.
If they are not held open, you can can just move them away with mv,
next CDR should just write new file.
Regards
A,tis
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On Thu, Jun 12, 2008 at 10:51 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup
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Cell
it can only be used for transfers.
Any ideas how i can solve my multiple cdr problem?
ResetCDR(w)
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask the caller the number they want to dial
+func+group
Calls will still be received by asterisk, however you will be able to
kick them off without proceeding with following dialplan logic.
Regards,
Atis
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-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
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will jump to h extension.
Regards,
Atis
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Cell Phone: +371 28806004
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every message mentioning Microsoft :p
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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,
and how I'm proposing.
Regards,
Atis
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/listinfo/asterisk-users
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for configs, etc. Imagine what will happen if that one PSU
will fail.
Regards,
Atis
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channels.
So, this should work with at least queue in ring-all mode (i feel that
it would be correct if Dial would do that too)
Regards,
Atis
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).
I suppose just a disconnect, because call was already bridged.
Regards,
Atis
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the call.
Any thoughts?
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a comment in bugtracker:
http://bugs.digium.com/view.php?id=12556
Regards,
Atis
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Cell Phone: +371 28806004
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in future
versions of Asterisk.
Regards,
Atis
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Atis Lezdins schrieb:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some action on
events like ringing
On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote:
However I encountered a resistance from Asterisk developers, as they
don't wish to accept my patches - because they don't wish to support
another interface. As I
to give full
overview of Asterisk Status.
Regards,
Atis
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in sip.conf
(mailbox= line).
Regards,
Atis
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, but I intend to do that in
future. It's been working stable on our production for several months.
If You're interested, please reply, and I'll try to separate that
patch out from other our patches.
Currently I have it updated for 1.4.19, but also have some version for 1.4.14
Regards,
Atis
--
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On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote:
--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Apr 28, 2008 at 8:34 PM, Vieri
[EMAIL PROTECTED] wrote:
How can I get a list of the callers within a
specific
queue at any given moment?
I need to get
Atis Lezdins wrote:
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably
/asterisk-users
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Atis Lezdins,
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to call '[EMAIL PROTECTED]'.
Giving up.
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Atis Lezdins,
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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on.
Maybe You replied to wrong topic?
Regards,
Atos
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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* and not for the cards themselves.
That's true.
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Atis Lezdins,
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[EMAIL PROTECTED]
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Cell Phone: +371 28806004
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: this is needed for a PBX connectect with a poor ADSL having
only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit)
and rejecting the 4th.
Any solutions?
if (${GROUP_COUNT([EMAIL PROTECTED])}) function in combination with
Set(GROUP(a)=x)
or Set([EMAIL PROTECTED])
Regards,
Atis
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Atis
type: friend
username: 21168
disallow:
allow: all
musiconhold: NULL
regseconds: 1207763735
ipaddr: 192.168.1.123
regexten:
cancallforward: yes
setvar:
call-limit: 4
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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the sound is done you start
MoH back up again. Probably a bit more involved than what you want,
but it dose work well for us.
MATT---
On 4/2/08, Atis Lezdins [EMAIL PROTECTED] wrote:
Sorry for top-posting, but seems everyone on this thread did so.
Also that would be my suggestion for now
Regards,
Atis
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Atis Lezdins,
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[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
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On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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to
allow combining of device states.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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caller and callee hangups. I
suppose dial time limit will match Callee hangup, but you can check
that by ${ANSWEREDTIME} or some sort of timestamp checking before and
after Dial (altough that would include ringing time)
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
application. If you have no intention to
use it, you might very well remove.
I've seen this problem once, however recompiling everything and
restarting helped me. I would suggest you just doing make clean on
zaptel and asterisk, then compile first zaptel, then asterisk.
Regards,
Atis
--
Atis Lezdins
asterisk.conf without sysname and create shell script:
#!/bin/bash
cat asterisk.conf.template
echo sysname=`hostname`.
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
nothing to worry about.
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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create_queue_member() function. This will allow you speed bonus
from hashtable in some places, and will make sure the login time gets
registred. You can also consider updating lastcall in
set_member_paused() - i'm having both of those.
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL
=MEMORY select * from cdr where dst =
4010 and calldate between 2008030800 and 20080313145900 group by
uniqueid;
and then compare:
SELECT * FROM a WHERE callid NOT IN (SELECT uniqueid FROM b)
SELECT * FROM b WHERE uniqueid NOT IN (SELECT callid FROM a)
Regards,
Atis
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Atis Lezdins,
VoIP
of ${CDR(UNIQUEID)}, but you can use just
${UNIQUEID}. If you want to pass variable to child channels, you
should make it inheritable. I'm using:
Set(__call_id=${UNIQUEID})
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
that in DB, manipulating CDR is the way to go.
When you will have more specific questions, please ask, i'm sure
somebody will answer :)
Regards,
Atis
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone
)
- end of log ---
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Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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