Re: [asterisk-users] show queue's name and other info in incoming call to queue member

belong to couple queues. Work around would be setting calling name with such information. If Your phone supports text CLID: Set(CALLERID(name)=${CALLERID(num) - Sales); Queue(sales); If not, You can just add some digit in front/end of CALLERID(num). Regards, Atis -- Atis Lezdins, VoIP

Re: [asterisk-users] show queue's name and other info in incoming call to queue member

On Thu, Dec 10, 2009 at 2:54 AM, Atis Lezdins a...@iq-labs.net wrote: On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys voi...@gmail.com wrote: hello,   I've callcenter and our queue members want to see on their IP phone's display queue's name , from which incoming call was originated

Re: [asterisk-users] queue issue

for the queue, not the amount of callers talking to queue members. You can do any limitations i can imagine with Set(GROUP()=...) and GROUP_COUNT. Do You actually need rest of callers to wait in queue while one is speaking, or disconnect them before they enter queue? Regards, Atis -- Atis Lezdins, VoIP

Re: [asterisk-users] Realtime with rtcachefriends=no problems...

only if it's string. You can safely do ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5); Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] application missed in asterisk 1.6.1 - SetCallerID()

() ... and maybe so more. anyone already notice that to ? If it's not normal, anyone have an solution to it ? Read the UPGRADE.txt Solution is to use functions instead: Set(CALLERID(name)); Set(CALLERID(num)); Set(CHANNEL(language)); etc Regards, Atis -- Atis Lezdins, VoIP Project Manager

Re: [asterisk-users] T.38 pass-through 488 handling problem

, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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

Re: [asterisk-users] T.38 pass-through 488 handling problem

On Mon, Jun 8, 2009 at 7:00 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Atis Lezdins schrieb: On Mon, Jun 8, 2009 at 2:06 PM, Klaus Darilionklaus.mailingli...@pernau.at wrote: Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38             Asterisk          ATA

Re: [asterisk-users] Question about core CDR system for multilpe servers

from specific system. However I would suggest not doing heavy SELECT's on this database, set up another slave for reports, as each table lock will cause asterisk posting a CDR to wait (and current call posting a CDR will wait in silence) Regards, Atis -- Atis Lezdins, VoIP Project Manager

Re: [asterisk-users] Queue - Multiple Transfer

, and put a Dial with t flag there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

Re: [asterisk-users] New tutorial: storing audio recordings per day

(|${TIMEZONE}|%Y/%m/%d)}); Set(MONITOR_FILENAME=${MONITOR_DIR}/${call_day}/call-${UNIQUEID}); Monitor(ulaw,${MONITOR_FILENAME},b); Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800

Re: [asterisk-users] Queue Load, Asterisk Disconnected

://www.voip-info.org/wiki/view/Asterisk+debugging - and compile asterisk without optimizations, make it crash and then look into core file with gdb. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone

Re: [asterisk-users] Support of /* */ comments in ael.vim

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

Re: [asterisk-users] Preferred language for Asterisk AGIs development ?

is troublesome unless You check internally for effective uid and call sudo internally. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Outgoing Queues

monitoring info which can get lost during restarts/reloads. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Record in mp3

Secondarily, MPEG audio compression takes a lot of CPU.  Until the last few years, desktop CPUs weren't even capable of doing realtime MPEG audio compression, which is necessary if you're going to have the recording ready by the time the audio input is terminated.  Above and beyond that, even

Re: [asterisk-users] [asterisk-dev] How to get to 10.000 open calls

://lists.digium.com/mailman/listinfo/asterisk-dev -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

functions. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

scripting. Why would the audio data path would be necessary? In our setup CallWeaver effectively acts as modem, and talks T.38 with provider. Please see my previous statement about desktop client software. I doubt that this can be simply achieved with custom scripting. Regards, Atis -- Atis

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

. Why would the audio data path would be necessary? In our setup CallWeaver effectively acts as modem, and talks T.38 with provider. Fax information data path to be pedantic. Data from Hylafax to CallWeaver is passed as TIFF image - thus no data/quality loss. Regards, Atis -- Atis Lezdins

Re: [asterisk-users] Exit Dial Application

on unanswered channel. You could try it opposite way - Dial from SIP phone to Zap. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

detailed descriptions of this mechanism. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth

Re: [asterisk-users] Ignoring time spent waiting in queue in CDR

requirements, and how they could change in future. Perhaps using the queue_log would allow rapid implementation and changes. Also, make sure to take a look at queue_log on Asterisk 1.6.0/1.6.1, they have some nice features added. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs

Re: [asterisk-users] Exit Dial Application

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, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone

Re: [asterisk-users] Ring All Queue

realtime queue log. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Exit Dial Application

, sorry, missed that part :) Try enabling full log in logger.conf, set verbosity to 3 and debug to 1, and see what goes in it. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689

Re: [asterisk-users] Exit Dial Application

in it. Check /var/log/asterisk/full (assuming default install location). You'll need to enable full line in logger.conf, restart Asterisk and issue core set verbose 3 and core set debug 1 in CLI. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net

Re: [asterisk-users] Exit Dial Application

for this Zap/ line? You could verify that by using Read before Dial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Asterisk is not designed for University with largeuser base?

from some interface, just issue sip prune realtime peer xxx trough manager. Also, in Asterisk 1.6 res_mysql driver can take advantage of MySQL master/slave setups, so You can distribute Your database load to separate read/write hosts. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer

Re: [asterisk-users] colorized logfiles in asterisk 1.6.0.6

); } return $result; } -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Bounty- CDR Bug Fix

://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, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1

Re: [asterisk-users] Bounty- CDR Bug Fix

a way to send it :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Meetme - play the name

is having DID (individual, unique across whole office), so this feature is called for. This is good reasoning for local users. The name prompt from voicemail could be used and made more generic. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype

Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

for those test calls. Also, thanks for showing us magics of ecasound. I have similar project (pbx-test-framework) that allows IVR/Queue/etc testing in automated mode. Recording everything and checking voice quuailty would be great addition :) Regards, Atis -- Atis Lezdins, VoIP Project Manager

Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

and fine. Sometimes i even log our production servers for weeks with debug 1. So i would suggest submiting this modification to digium bugtracker, if it really helps tracking ip's. Thanks again, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net 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

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

will be next to get call, but not which call will be sent to next agent (if i understood OP correctly) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

will have an effect on the order that calls are picked up. Yes, announcments could also affect this. If announcement is being played to caller, he won't get connected at that point, and other call could jump in front of him. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ

Re: [asterisk-users] Asterisk 1.4.22 Queues problems (Fifo or not ?)

On Thu, Dec 18, 2008 at 9:44 PM, Benoit maver...@maverick.eu.org wrote: Atis Lezdins a écrit : On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw dhens...@ignition.bm wrote: I believe you are correct Atis. Philipp within your queue setup do you have any announcements? If so read the posting

Re: [asterisk-users] Asterisk / Hylafax

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

Re: [asterisk-users] Asterisk / Hylafax

/pipermail/asterisk-users/2008-November/222531.html Please mind, that if you're trying T38modem, you should get versions exactly as specified in voip-info.org, otherwise they might not work with Opal (which adds SIP protocol, as T38modem was originally for H.323) Regards, Atis -- Atis Lezdins

Re: [asterisk-users] 1.6 upgrade issues

,provider2); Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Asterisk spoken digits

://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

Re: [asterisk-users] config from DB

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

Re: [asterisk-users] CDR Design

, 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

Re: [asterisk-users] CDR Design

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

Re: [asterisk-users] top posting again [was: Re: CDR Design]

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

Re: [asterisk-users] top posting again [was: Re: CDR Design]

/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

Re: [asterisk-users] CLI and choice of messages

) 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

Re: [asterisk-users] top posting again [was: Re: CDR Design]

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

Re: [asterisk-users] Parking calls

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

Re: [asterisk-users] Parking calls

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

Re: [asterisk-users] Parking calls

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

Re: [asterisk-users] Parking calls

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

Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released

: 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

Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released

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

Re: [asterisk-users] Priority between calls from different queues

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

Re: [asterisk-users] Priority between calls from different queues

, 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

Re: [asterisk-users] Priority between calls from different queues

, 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

Re: [asterisk-users] Any 1.6 SendFAX example ?

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

Re: [asterisk-users] Any 1.6 SendFAX example ?

(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

Re: [asterisk-users] SVN

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

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

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

Re: [asterisk-users] SendImage()

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

Re: [asterisk-users] Asterisk 1.6 mysql cdr log problem

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

Re: [asterisk-users] Log level of 500 Server Internal Error.

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

Re: [asterisk-users] database queries from extensions.conf

__ ___ -- 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

Re: [asterisk-users] Ping

://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

[asterisk-users] Log level of 500 Server Internal Error.

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

Re: [asterisk-users] Log level of 500 Server Internal Error.

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

Re: [asterisk-users] Limit the number of users in a meetmeconference?

() 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

Re: [asterisk-users] Any other free toll free SIP providers out there?

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

Re: [asterisk-users] Macro conversion in 1.6

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

Re: [asterisk-users] IF else

-- 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

Re: [asterisk-users] IF else

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

Re: [asterisk-users] Debugging Asterisk

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

Re: [asterisk-users] How long will Asterisk 1.4.x supported/maintained

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

Re: [asterisk-users] Looking for a good lightweight Linux softPhone

-- 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

Re: [asterisk-users] RTP LOG

/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

Re: [asterisk-users] Looking for a good lightweight Linux softPhone

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

Re: [asterisk-users] QueueLog from AMI

. 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

Re: [asterisk-users] QueueLog from AMI

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

Re: [asterisk-users] QueueLog from AMI

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

Re: [asterisk-users] List eating mail again?

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

Re: [asterisk-users] tired of midget packet received warnings

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

Re: [asterisk-users] Variable Scope Question

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

Re: [asterisk-users] [OT] Capitalism (was: Spam from DIDForSale [EMAIL PROTECTED])

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

Re: [asterisk-users] AEL NoOp not working

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

Re: [asterisk-users] Phishing attempt

?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

Re: [asterisk-users] AEL NoOp not working

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

[asterisk-users] [OT] Flash player for call recordings - 8khz

-- 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

Re: [asterisk-users] Fring: Open VPN client to be installed on the mobile, which mobile?

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

Re: [asterisk-users] Agents log in afterhours

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

Re: [asterisk-users] [help] Realtime Swich any context dinamically

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

Re: [asterisk-users] a little regex help needed

-- 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

Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4

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

Re: [asterisk-users] Asterisk voicemail

, 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

Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4

(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

Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4

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

Re: [asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'

, 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

Re: [asterisk-users] Question on using DMZ

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

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