___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis
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
://lists.digium.com/mailman/listinfo/asterisk-users
--
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
___
-- Bandwidth and Colocation
everything that's available from web.
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
___
-- Bandwidth and Colocation
=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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
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
___
-- Bandwidth
unsuccessful on that part.
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
___
-- Bandwidth and Colocation Provided by http
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype
of call. A little clutter, but it works more or less.
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
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
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
[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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype
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
--
Atis Lezdins,
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
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
from web server to Asterisk server.
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
___
-- Bandwidth and Colocation
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
| MEMBERALREADY | NOSUCHQUEUE
Example: AddQueueMember(techsupport|SIP/3000)
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
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
--
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
reload.
As for executing CLI commands, see manager action Command:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1
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
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
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
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
--
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
, 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
--
Atis Lezdins,
VoIP Project Manager
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
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
.
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
___
-- Bandwidth and Colocation Provided by http://www.api
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
app_queue.so would do the trick :)
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
___
-- Bandwidth and Colocation
had passtrough mode and 1.6 can send and receive.
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
___
-- Bandwidth
.
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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
-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
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
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
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
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
,
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
INSERT INTO cdr_log ...
Is there anyone who can help me?
--
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
___
-- Bandwidth
, there's command -t which could be passed at asterisk
startup, then asterisk will write all files in /var/spool/asterisk/tmp
(allocating empty filename before), and after recording finishes it
will move them to correct location.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer
(qlog, %ld|%s|%s|%s|%s|,
(long)time(NULL), callid, queuename, agent, event);
[...]
+ }
}
-Original Message-
From: Atis Lezdins [mailto:[EMAIL PROTECTED]
Sent: Monday, 13 October 2008 8:02 PM
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non
Configuration Driver 0
1 modules loaded
This should also be fine.
You could also try catching me on irc, just look for atis_work or
atis_home in #asterisk.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
--
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
you would need to issue dialplan reload or
AEL reload whenever you add a context.
Regards,
Atis
P.S.
try to not post twice :)
--
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
after hours all agents are logged out every 15 minutes. So,
they are allowed to work after official working hours, but they just
have to relogin every 15 minutes. Realtime queue members in MySQL and
cron script makes this quite straightforward :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project
and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
--
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
something obvious ?
Hi,
NoOp is not outputting anything, it's just does nothing, however you
should still be able to see Executing NoOp(blablabla) in console,
as it's a command.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone
?RGlnaXVt?= [EMAIL PROTECTED]
--
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
___
-- Bandwidth and Colocation Provided by http://www.api
On Wed, Nov 5, 2008 at 5:28 PM, Olivier [EMAIL PROTECTED] wrote:
2008/11/5 Atis Lezdins [EMAIL PROTECTED]
On Wed, Nov 5, 2008 at 12:39 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2
I'm using NoOp and Verbose functions inside
the call will go (within
Asterisk of course) you will have variable ${company}
For more information please see http://www.voip-info.org/wiki-Asterisk+variables
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
recently submitted idea for Google Project 10^100 which would help
implementing Resource Basec Economy (i just didn't knew that such term
exists). Can't wait January 27th.. :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371
the same flexibility. You can disable
specific log levels (for example warnings) in logger.conf or you can
log everything to syslog, where filter out this specific message.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
.
Regards,
Atisw
--
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
backporting 3 added lines) when
upgrading to 1.6.1.
http://svn.digium.com/view/asterisk?view=revrevision=120166
Regards,
Atis
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 3:16 PM
Para: Asterisk
in month or two. Next release in 1.6.0 branch will be
1.6.0.2.
Regards,
Atis
Regards
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins
Enviado el: Wednesday, November 12, 2008 5:12 PM
Para: Asterisk Users Mailing List - Non-Commercial
Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
--
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis
On Fri, Nov 14, 2008 at 10:27 PM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Fri, Nov 14, 2008 at 08:34:48PM +0200, Atis Lezdins wrote:
On Fri, Nov 14, 2008 at 7:07 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
On Fri, 14 Nov 2008, Gordon Henderson wrote:
On Fri, 14 Nov 2008, Tilghman
there.
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
___
-- Bandwidth and Colocation Provided by http://www.api
tag (for example
1.4.19 to 1.4.22)
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
___
-- Bandwidth and Colocation
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Wed, Nov 19, 2008 at 6:51 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID
they could even pay for advertising to get
included there ;-)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
was
changed from pipe to comma. Unless you read it, you might also
experience lot of other problems.
It should be Macro(phones,200,SIP/200)
However it's not recommended to use macro's, you are encouraged to
convert them to GoSub's, as they now support arguments.
Regards,
Atis
--
Atis Lezdins,
VoIP
://lists.digium.com/mailman/listinfo/asterisk-users
Pong
GMail's preview looks fun - Ping -- Bandwidth and Colocation Provided
by http://www.api-digital.com;
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
checking logs for warnings and errors, so i probably
have missed those.. It would be great indication that something is not
ok - either outgoing trunk or local phone is bad.
Any opinions?
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype
On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i sometimes get those back from provider. They are
currently general SIP message log entries
() in the dialplan. Thanks for the info!
- Noah
If it's in realtime, then it should also work from config file. If
it's not, then file a bug.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone
is
insignificant, nobody should be offended..
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth
should detect
table structure and warn about missing fields. If it's so, perhaps you
can change asterisk - mysql (res_cdr_addon_mysql if i remember
correctly) to do an alter on your table - then it will automagically
create missing fields.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager
On Fri, Nov 21, 2008 at 7:48 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Fri, Nov 21, 2008 at 7:32 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
Hi,
VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error
I just noticed that i
__
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager
On Tue, Nov 25, 2008 at 2:19 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 24 November 2008 11:38:09 am Atis Lezdins wrote:
On Sun, Nov 23, 2008 at 2:19 PM, Artifex Maximus [EMAIL PROTECTED]
wrote:
I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers
and tools but my
info about how it's not working for you.
Probably it's that http://svn.digium.com/ gives 403 error.
As i recall, it showed up when some search engine tried to indexing
whole SVN ignoring robots.txt, so Digium disabled root page. Now you
can access it by adding /view/ to URL.
Regards,
Atis
--
Atis
dial destination number (SIP/[EMAIL PROTECTED]) and send
local side of channel to fax_out,${NUMBER},1 which does SendFax.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work
(which sends
trough Asterisk with T38 passtrough).
So, if you have PRI ir analogue lines, use IAXmodem, otherwise you
have to do either T38modem or SendFax.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
callers within queue
by setting QUEUE_PRIO variable before sending call to queue.
You could try to describe why you need two queues and what should be
rules to distribute calls - so we can help you with overall
architecture.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs
, Atis Lezdins [EMAIL PROTECTED] wrote:
On Fri, Nov 28, 2008 at 1:13 PM, equis software [EMAIL PROTECTED]
wrote:
Hi!
I want to know the way that calls are answer in this case...
I have queue1 and queue2, one agent that receive call from both queues.
queue1 - call1
queue1 - call2
queue2 - call3
, but it could be complex :)
Regards,
Atis
regards
On Fri, Nov 28, 2008 at 12:31 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Fri, Nov 28, 2008 at 4:16 PM, Darrin Henshaw [EMAIL PROTECTED]
wrote:
One thing you also will run into is listed here:
http://www.voip-info.org/wiki/view
: error: (Each undeclared identifier is reported only once
manager.c:1732: error: for each function it appears in.)
make[1]: *** [manager.o] Error 1
make: *** [main] Error 2
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell
2008 i686 GNU/Linux
Debian Sid - Linux debian 2.6.26-1-686 #1 SMP Thu Oct 9 15:18:09 UTC
2008 i686 GNU/Linux
1.6.0.1 compiled fine on at least two Fedoras.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
to IVR or even Dial() and at
later point check results. For example you can add G or M argument to
Dial() to execute part of dialplan macro/gosub upon answer.
Hope that my explanation helps :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype
it shouldn't be a problem. All you
need is to store ${CHANNEL} name of current channel before entering
MusicOnHold().
Also you could take a look at GROUP_COUNT function, perhaps it in some
way can help you :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc
it worked!;
}
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: miércoles, 03 de diciembre de 2008 03:48 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking calls
On Wed, Dec 3
spitted out ideas of
how i would solve it. I looked at available commands, and if you say
MusicOnHold doesn't stop, you have to terminate it somehow.
Regards,
Atis
Thanks for your solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
, but it's hard to find time for reading RFC
(i'm in middle yet). So, i hope this will go on and allow me to
respond with some objective comments.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1
refrain until i
complete reading Murf's RFC. I just don't feel competent enough to
speak about this without reading he's ideas first.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
policy
It's not official policy, however it's pleasant in long discussions.
It's good to make it a personal habit :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work
/asterisk-users
i have the solution so every one is happy i will write over and below :- )
əsuəs səʞɐɯ
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
) and it will be printed
out with Verbosity of 0. That's default verbosity you see in CLI.
NoOp really does nothing as opposed to Verbose(), so you will see it
only in -- Executing message which has verbosity 2.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL
reader which will automatically scroll to the top
of the latest info, let me know. If there is a technological fix,
perhaps these threads will die down.
GMail webinterface does automatically hides quotations. I expect that
other mail clients are following.
Regards,
Atis
--
Atis Lezdins,
VoIP
jumping to other context. Upon returning from gosub it would be back
the same.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
://www.voip-info.org/wiki/view/Asterisk+cmd+SetLanguage
Set(CHANNEL(language)=my)
and put your digits in /var/lib/asterisk/sounds/my/digits
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1
installing ffmpeg of course. Local
copies of opal i have mentions libavcodec/ffmpeg only in plugins dir.
Did you compiled plugins? Perhaps you can try deleting everything
there.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone
101 - 200 of 330 matches
Mail list logo
