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 wrote: > On Mon, Dec 7, 2009 at 10:00 AM, Giedrius Augys 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

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

ears when one member > can 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).

Re: [asterisk-users] queue issue

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 Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net S

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

ge() ... 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 Lez

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

ea ? Asterisk Realtime Architecutre currently treats all fields as strings. I wish too that it would take into account actual field type retrieved from DESCRIBE statement and add the quotes only if it's string. You can safely do ALTER TABLE sip_buddies CHANGE COLUMN port port VARCHAR(5); Regards,

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

On Mon, Jun 8, 2009 at 7:00 PM, Klaus Darilion wrote: > > > Atis Lezdins schrieb: >> On Mon, Jun 8, 2009 at 2:06 PM, Klaus >> Darilion wrote: >>> Hi! >>> >>> I have the following problem with Asterisk 1.4.23: >>> >>> >>> AT

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

teway app (don't remember if there exists any and in what state), or just write a RxFax which would then generate call with TxFax. 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] Question about core CDR system for multilpe servers

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 / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806

Re: [asterisk-users] Queue - Multiple Transfer

and set TRANSFER_CONTEXT variable, 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 _

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

is owner of parent directory. Set(__call_day=${STRFTIME(|${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.lez

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

; "0227559600" <0227559600>") in new stack > -- Executing Set("Local/2...@from-internal-a118,2", "FROMCONTEXT=exten-vm") > in new stack > -- Executing Macro("Local/2...@from-internal-a118,2", "record-enable|225|IN") > in new st

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

On Mon, May 11, 2009 at 1:55 PM, Philipp Kempgen 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,

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

e 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 Wor

Re: [asterisk-users] Outgoing Queues

e's queue_log for that. This is purely 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 80

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, ev

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

://lists.digium.com/mailman/listinfo/asterisk-dev > > > _______ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit:

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

when the same could be >> > achieved using Callweaver alone and some custom scripting. >> >> 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.

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

d when the same could be achieved using > Callweaver alone and some custom 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 ca

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

ify whole setup when migrating to Asterisk 1.6, which would take over CallWeaver 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

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

ed to execute custom scripts that grab generated .tiff files and feed them to CallWeaver. Just search list archives, I've writen detailed descriptions of this mechanism. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ

Re: [asterisk-users] Exit Dial Application

7;d' implies an answered channel? Or is this a Bug? > I think the limitation could be by analogous Zap phones, as they probably don't support sending DTMF on unanswered channel. You could try it opposite way - Dial from SIP phone to Zap. Regards, Atis -- Atis Lezdins,

Re: [asterisk-users] Exit Dial Application

w, does DTMF work at all 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

Re: [asterisk-users] Exit Dial Application

ging That's CLI interface output, log should have timestamps and much more detail 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 "

Re: [asterisk-users] Exit Dial Application

rk, but it > doesn't : / > Oh, 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 Ce

Re: [asterisk-users] Ring All Queue

quot; for that. Of course, if You need it only on hangup, Luis suggestion will work just fine, use Asterisk Realtime engine to read value from realtime queue log. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +

Re: [asterisk-users] Exit Dial Application

ny.at > -BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (MingW32) > > iEYEARECAAYFAknkvqoACgkQR0exH8dhr/ZqRACfV7KLoTMl9RgH0QNIPiJ/Gq9G > 5dcAoIVK3L7pxTBLZrDi+kJGpOCPVa47 > =hEGE > -END PGP SIGNATURE- > > ___ >

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

ver You should really have a think about what are Your 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

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

user credentials 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 Le

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

html($str) { $tokens = explode(chr(27).'[',$str); $result = array_shift($tokens); foreach ($tokens as $k=>$v) { $end = 8; $code = substr($v,0,$end); if ($code=='0;37;40m') $result .= ''; else if ($code=='1;36;40m') $result

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

ting a bounty out on it. > > http://bugs.digium.com/view.php?id=13691 > > > > > > I would not recommend using CDR's for queue data, instead I use the > queue events, or at a minimum the queue log. > > > > > ---- > >

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

e out 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 Colocati

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

> tell them apart based on callerid. >In my case, every person 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

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

7;re using and everything should work fast 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 / Develope

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

able to keep track of their billing, etc 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 :) Regard

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

On Thu, Dec 18, 2008 at 9:44 PM, Benoit wrote: > Atis Lezdins a écrit : >> On Thu, Dec 18, 2008 at 8:50 PM, Darrin Henshaw wrote: >> >>> I believe you are correct Atis. >>> >>> Philipp within your queue setup do you have any announcements? If so read

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

ments 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 /

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

affects which agent 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

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

ting it in some part of dialplan. 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 C

Re: [asterisk-users] 1.6 upgrade issues

> should be along the lines of: Gosub(outbound,s,1 > (${EXTEN},provider1,provider2)). > Actually there's ampersand operator prefixing macro name, so AEL parser will automatically check dependencies etc: &outbound(${EXTEN},provider1,provider2); Regards, Atis -- Atis Lezdins, VoIP Pr

Re: [asterisk-users] Asterisk / Hylafax

c on start up? I'm really not sure. You can try 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

Re: [asterisk-users] Asterisk / Hylafax

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

Re: [asterisk-users] Asterisk spoken digits

om files in one location. http://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 P

Re: [asterisk-users] config from DB

(${CALLERID(num)}) to it. Remember that ${EXTEN} is just any number in your dialplan, and you can set it to CallerID when 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

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

at is new. > > If you know of a mail 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 client

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

hing) with Verbose(something) 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

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

no less required. >> >> -- >> Tilghman >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: &g

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

t you place your reply here? > > We have archives of the list. We can spot the original message. > > [snip more useless quoting resulted from top-posting] > > > > Sorry I did not know you have a non-top-posting policy > > It's not official policy, however it's pleasant

Re: [asterisk-users] CDR Design

n stays on that call for a long time - > who's picking up the bill? > > Current CDR's are lacking in this respect - and I think this is what > murf is trying to sort out (please jump in here murf). > I would like to comment really much of this, but I'll refrain until i c

Re: [asterisk-users] CDR Design

SIP registrations etc). > > I guess we'll just have to wait and see what santa murf gives us all for > Christmas :). > I really want to contribute this discussion (and RFC), i'm reading it and i have lot of to say, but it's hard to find time for reading RFC (i'm

Re: [asterisk-users] Parking calls

-- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins > Sent: miércoles, 03 de diciembre de 2008 10:31 p.m. > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Parking calls > > On Thu, Dec 4, 2008 at 1:25 AM,

Re: [asterisk-users] Parking calls

t; that not involves agi. >> >> Any idea?? >> >> > > AMI action Redirect - > http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect > > Of course you would need some script to send this action, but as long > as you control writes to databas

Re: [asterisk-users] Parking calls

ct Of course you would need some script to send this action, but as long as you control writes to database 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, perha

Re: [asterisk-users] Parking calls

ry etc. If you just have to do something heavy for each call and you don't use result of that operation to determine next step of call, you can do: System((/usr/bin/do-something.sh)&) note, the ampersand after first brackets will make to run shell command in background. I

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

DT 2008 x86_64 x86_64 x86_64 GNU/Linux Debian Etch (4.0) - Linux saule 2.6.18-6-xen-686 #1 SMP Thu May 8 11:28:36 UTC 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 -- Ati

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

anager.o manager.c: In function 'action_getvar': manager.c:1732: error: 'SENTINEL' undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears in.) make[1]: *** [manager.o] Erro

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

wer channel, even if you set "r" option.. not sure is this a problem, 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, Darri

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

from queue2 - no matter that queue2 has lower weight or whatever settings. To overcome this, you have to enable shared_lastcall (available since 1.6.0). Regards, Atis > > Regards > > On Fri, Nov 28, 2008 at 11:34 AM, Atis Lezdins <[EMAIL PROTECTED]> wrote: > > On Fri, Nov 28

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

multiple calls are ready to go to agent in different queues. Also, you can give priority to different 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 yo

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

and generates call file for Callweaver (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: at

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

estination,$vars,$callerid,$waittime,$deliver_time,$filename,$retries,$callfile_dir); Of course you'll need ast_originate_callfile which writes data to file and then moves to correct dir. I would publish that, but it's full of my constants and realted to much other libs.. Basically,

Re: [asterisk-users] SVN

bled root page. Now you can access it by adding /view/ to URL. 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] 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

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

___ > This email has been scanned by the MessageLabs Email Security System. > For more information please visit http://www.messagelabs.com/email > __ > > __

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, >>>> >>>> VE

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

and 1.6 log system. > You should also check Asterisk log for warnings. 1.6 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 a

Re: [asterisk-users] SendImage()

behavior. Current users > see an issue either way, and future users won't see a problem at all. > Perhaps somebody from -dev team can be delegated to check naming consistency of new features? So, whenever a feature is added (perhaps at code review), he checks naming to match best of he&#x

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

tabase and enforced >> on RealTime enable conferences. This presumes you are >> looking at 1.6.X or Trunk code... > > Ah. No realtime for me, so I guess I'll just stick with using > MeetmeCount() in the dialplan. Thanks for the info! > > > - Noah > If it

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 provid

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

rently i'm 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 P

Re: [asterisk-users] Ping

ate options visit: > http://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

Re: [asterisk-users] Macro conversion in 1.6

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 Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.

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

+Termination+Providers So, now it's updated with FWD and IdeaSIP, and linked from "VoIP Service Providers" Perhaps anyone who uses them can check examples - the ${EXTEN:1} part seems wrong. I wonder are there any legal issues if they were included in Asterisk sample config? O

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. => {

Re: [asterisk-users] IF else

w,,5) exten => _111,n,Wait(2) exten => _111,n,Playback(/tmp/asterisk-recording) exten => _111,n,Wait(2) exten => _111,n,Hangup exten => _112,1,Noop(Dialed 112) exten => _112,n,Playback(AR_GetGiveToID) exten => _112,n,Wait(2) exten => _112,n,Record(/tmp/asterisk-recording:ulaw,,

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

e that's what i usually do. And then there's also SVN switch, to update to other tag (for example 1.4.19 to 1.4.22) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phon

Re: [asterisk-users] Debugging Asterisk

ll is, and then do a "zcat" on compressed logs. Also, i've heard that this approach of one uniqeid for all child channels has been committed in trunk, it's called "linked_id" there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTE

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

Re: [asterisk-users] RTP LOG

update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-user

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

gt; tool you will need an asterisk server to connect to to place your calls. > I am not understanding where you think the bloatware is coming into play. > > So are you sitting at the console of the machine running asterisk or is > this something that you would use from a standalone *nix wor

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

ial Liberty to purchase a little Temporary > Safety, deserve neither Liberty nor Safety." > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailma

Re: [asterisk-users] QueueLog from AMI

so it won't be out 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 P

Re: [asterisk-users] QueueLog from AMI

g it in mind (if not even backporting 3 added lines) when upgrading to 1.6.1. http://svn.digium.com/view/asterisk?view=rev&revision=120166 Regards, Atis > > > -Mensaje original- > De: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] En nombre de Atis Lezdins > Enviado

Re: [asterisk-users] QueueLog from AMI

vent, "%s", message); So, agent would be "Interface" and data would be "Message". However, i wonder why do you need to pass "Login" event, as any kind of Queue Login (dialplan or AMI) would do that automatically. Regards, Atisw -- Atis Lezdins, VoIP Project

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

> And what if you can't fix the source of these packets? And what if > friendly peers outside of your realm (likely to iax-call you, so can't > block them) sends these packets? There are holes in your logic. > > So asterisk has to be puritan of the lot? Holier than thou? Pro

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

e do, go back to a >> barter economy? :-) >> >> Thanks for interesting link :) Didn't knew any such projects exist. I recently submitted idea for Google Project 10^100 which would help implementing Resource Basec Economy (i just didn't knew that such term exists). C

Re: [asterisk-users] Variable Scope Question

execute Set(__company=A). Two underscores means that this variable will be inherited in every child channel, so wherever 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 -

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:/

Re: [asterisk-users] Phishing attempt

EMAIL PROTECTED] designates 66.48.80.65 as permitted sender) [EMAIL PROTECTED] ReturnPath: "Digium"<[EMAIL PROTECTED]> X-Mailer: SMTP Message-ID: <[EMAIL PROTECTED]> MIME-Version: 1.0 From: =?utf-8?B?RGlnaXVt?= <[EMAIL PROTE

Re: [asterisk-users] AEL NoOp not working

e while Verbose > is working. > Am I missing 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, At

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

r [2] http://www.varal.org/media/niftyplayer/ 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 Co

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

ion 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 >> >> >>

Re: [asterisk-users] Agents log in afterhours

ing might send call to IVR first to welcome caller and give agents some time. 2) Within 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 mak

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

a script that would do "SELECT DISTINCT context FROM extensions_table" and for each of results print out "switch=>" lines. However you would need to issue "dialplan reload" or AEL reload whenever you add a context. Regards, Atis P.S. try to not post twice :) --

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

's also direct callerid matching, so you can match dialed extension and callerid in same rule, but this looks simpler to me in this case :) For more info see http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf and search for "ex-girlfriend" :) Regards, Atis -- Atis Lezdi

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

ng there. To debug >> this you shouldn't need more than "core set verbose 3" and "core set >> debug 1". > > I turned on debug mode and tried an agent login and logoff. > However, when I looked into debug and messages, there are lots of > chan_sip.c and a few cdr_add

Re: [asterisk-users] Asterisk voicemail

y have temporary storage in /tmp/, however there's more general option for asterisk. See "man asterisk", 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 re

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

"event", event, > + "data", qlog_msg, > + NULL); > + } else { > + if (qlog) { > + AST_LIST_LOCK(&logc

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

Id CommandArgument > 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 > 2 Connect [EMAIL PROTECTED] on db1 > 081013 16:00:32 1 Query INSERT INTO cdr_log ... > 081013 16:01:42 1 Query INSERT INTO cdr_log .

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

iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work p

Re: [asterisk-users] Question on using DMZ

> http://lists.digium.com/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:/

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