Re: [asterisk-users] check if not human

/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 -- Ing. Miguel

Re: [asterisk-users] real time connection failed

. :-) BTW, I think one could do that without restarting asterisk, simply enabling the debug in logger.conf and doing a logger reload, the 's with core set verbose X and the 's with core set debug X on the CLI. Please correct me if I'm wrong. Philipp Kempgen -- Ing. Miguel Molina

Re: [asterisk-users] HP DL380 G5 with TE420

. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Queue log on MySQL realtime

you can have several backends and still save the CDR on a plain text file? This could be useful to have a backup in case of a MySQL malfunction, loss of connection or something. For examples, the plain CDR file has been useful to me on some undesired data loss situations. TIA. -- Ing. Miguel

Re: [asterisk-users] chan_zap.so missing

-- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Script to softly restart Asterisk each midnight to clean locked channels

version and cherrypicking patches that may or may not resolve the issues that we could experience. Just my 2 cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587

[asterisk-users] ${UNIQUEID} variable and queue log issues on 1.4.22

. Should I post bug notes about this issues? Any comments welcome. Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth

Re: [asterisk-users] DISA

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth

[asterisk-users] Know who's logged in

on this, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] Know who's logged in

on this, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Know who's logged in

Mr. James W. Laferriere escribió: Hello Mark , On Fri, 27 Mar 2009, Mark Michelson wrote: Mr. James W. Laferriere wrote: On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin

Re: [asterisk-users] Queues in memory after startup

, some of the changes made to trunk to improve IAX2 performance, and some other big changes are just awesome. Thanks. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587

Re: [asterisk-users] 400 calls at g711 how much cpu power

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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57

Re: [asterisk-users] Asterisk G729 codec...

mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587

Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk (update)

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 -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1

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

/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Dear asterisk-users@lists.digium.com PharmacyOnline Sale 79% OFF!

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Simultaneous Calls at a time

: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Sequential Ring Groups?

/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] Asterisk process ended

/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] Asterisk process ended

___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología

Re: [asterisk-users] CDR issue

___ -- 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 -- Ing. Miguel Molina Grupo de Tecnología

[asterisk-users] From 1.4 to 1.6.0

. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] play with varibles

: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] FW: Writing Hangup causes to CDR record

://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Maximum cable length for analog phone from FXS port

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth

Re: [asterisk-users] Suddenly the voice became garbage (like robot)using Asterisk 1.4.19.2

. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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

[asterisk-users] CPU usage vs compiler flags

solved for this and other issues, specially chan_local ones which we use a lot, so basically is to benefit of 6 months of bug fixes. Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation

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

in a single database server, shouldn't be a problem. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] CPU usage vs compiler flags

Tzafrir Cohen escribió: On Mon, Jun 01, 2009 at 09:23:48AM -0500, Miguel Molina wrote: Hi all, I just upgraded a production server to asterisk 1.4.25, compiling with the following: [*] 1. DONT_OPTIMIZE [*] 2

Re: [asterisk-users] Teliax: where's the space in CALLERID(num) from?

extens...@context . After you see what happens with the call, you can hangup it with the CLI hangup command if the dialplan doesn't hangup it already. Also, if your PC or server has a well configured sound card and speakers, the console call would be heard too. Regards, -- Ing. Miguel Molina

[asterisk-users] Help with asterisk core dump

) optimized values is marked as useless information, but IMHO it still helps to know what's failing inside it. I appreciate any input about this, could be this a bug? A library problem? Or a server memory problem? Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone

Re: [asterisk-users] Help with asterisk core dump

Matthew J. Roth escribió: Miguel Molina wrote: I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib

Re: [asterisk-users] Call recording in - out

off-hours. That was years away, before the MixMonitor() application bugs was fixed on 1.2 and it's quite stable since then. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] T38 support

. For the exact configuration parameters for a host, google is your friend :) Cheers, Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation

Re: [asterisk-users] Dialer program

Carlos Ruiz Diaz escribió: I can't find GNUDial web page :( It looks like the www.gnudialer.org is down. However, the sources are still in the same place: http://dynx.net/ASTERISK/gnudialer/ -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] multiple PRI's in one group ..how??

group=7 signalling = pri_cpe context=inbound-group channel = 1-5 This way, if you Dial group 1, you won't use channels 1-5, leaving them free for inbound calls. Cheers, Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] 1.6.0.10: core restart on ReceiveFax()

to receive simultaneous faxes at a time. After the change my machine is rock stable, receiving and sending hundreds of faxes a day with no restarts. Hope this helps. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] Asterisk + mySQL

add what Tilghman told you into modules.conf. cdr_mysql.conf is for MySQL CDR backend database settings. res_mysql.conf is for MySQL Asterisk Realtime Architecture (ARA) backend database settings. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] PRI cause codes

hand, have you tried changing the switchtype from national to euroisdn? It may help if your telco uses that type of switches, because the call setup info varies a little on each configuration. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

[asterisk-users] Persistent dynamic queue members

? Is there a way to workaround it? - Is someone else running into this? - Should I report this in the bugtracker? Thanks in advance, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Persistent dynamic queue members

Miguel Molina escribió: Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid

Re: [asterisk-users] MixMonitor/Queue and Tranfers

in case of a transfer, because the callee leg changes but the caller leg is the same. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] asterisk addon mysql - is mysql connection persistent

or you killed the connection from the MySQL monitor. So if for any reason you need to do a quick restart of MySQL, you won't lose CDR records if no calls are hungup during the MySQL restart cycle. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] MixMonitor/Queue and Tranfers

Un-topposting... On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co wrote: Darrin Henshaw escribió: 2. The issue does seem to be limited to MixMonitor and the Queue application, as in testing I setup mixmonitor on my extension dialed it from outside

Re: [asterisk-users] Queue autopause

my two cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

Re: [asterisk-users] Queue autopause

would imply a external application that monitors the queues and makes the custom autopause you need. Just my two cents... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] beeping in headsets from queue callers

Darryl Williams escribió: How do I turn off the beeps in the head sets when customers are waiting in the Queue? Look for the option announce in queues.conf. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] Originate (Executing a System Command)

this? Did you reload it and check it within the CLI? [dummy] exten = 1,1,Answer() exten = 1,n,Wait(2) exten = 1,n,Hangup() The application (your external command) won't be launched until the Origiante channel is answered. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] T38 negotiation, the last step !

, this message shouldn't be a problem. I think this is much like the X-lite 126 codec type message: [Jul 16 11:35:57] NOTICE[10989]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0' And doesn't pose a problem anyway. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] quenstion about asterisk

' ,'s', 'h', and others please RTFM: http://downloads.oreilly.com/books/9780596510480.pdf -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] Delete voicemail after couple of days

... Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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] *?

my subject :S Not all people (certainly more in this list) are expected to be ultragigageeks. Have a nice day. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Delete voicemail after couple of days

this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. Aloysius Thevarajah Lloyd escribi�: you want me to delete all the sytem files:) On Fri, 17 Jul 2009, Miguel Molina wrote: Yeah he wants to make yourself silently blow your

Re: [asterisk-users] Channel Variables in a Call file?

(seconds) on your dialplan to keep the channel alive while you check its info with the command. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] Count Available Queue members

, just let me know with the asterisk version you use. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Goto from a feature macro is not working?

of the better GoSub() one. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Goto from a feature macro is not working?

Philipp Kempgen escribió: Miguel Molina schrieb: I just ran into a similar problem, I needed a macro spreaded over several contexts because it's kind of a part of an IVR. I switched to GoSob() and Return() applications (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Gosub

Re: [asterisk-users] Fax for Asterisk quick question

, having that faxing is traditionally slow. So, it's pretty much a little more elaborate answer of unlikely to be pleasant. A direct T.38 gateway for asterisk would be awesome in any case of use. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] AMI not show originate on CLI

: Maybe in the debug log you might surely get something, the debug throws a lot of info from asterisk. Don't let the debug log grow too big. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation

Re: [asterisk-users] SIP vs Analog lines

legacy. The last time I put a T1 channel bank into use was more than two years ago, and never had to configure another one since then. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] SIP vs Analog lines

Steve Totaro escribió: On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote: John F. Ervin escribió: Never having actually rolled an Asterisk (Trixbox in my case) system into production. I was wondering if in most

Re: [asterisk-users] Dialplan SIP call back problem

. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list

Re: [asterisk-users] CDR Problem - No CDRs when call is not bridged

= 999,1,Playback(tt-weasels|noanswer) exten = 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not for 999. How can I change this behavior? Thanks Klaus Try unanswered = yes on cdr.conf Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] res_speech_lumenvox.so: undefined symbol: ast_speech_register

some module dependencies issue. What's the output of ldd /usr/lib/asterisk/modules/res_speech_lumenvox.so? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Server linux requirements

. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing

Re: [asterisk-users] Inbound Call coding

the advantages of web development, the old propietary PBX activity codes seem to be obsolete nowadays. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon

Re: [asterisk-users] A problem with recoding agents calls via monitor

kind? Or are you checking the plain CSV CDR file? Supposing that you have a MySQL backend, you should have the userfield=1 setting in cdr_mysql.conf to tell the backend to save the userfield of the record. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] Creating an ISDN PRI-to-SIP/IAX2 gateway

ISDN PRI to SIP or IAX2 gateway. Modify the dialplan patterns according to your needs. For your PRI zaptel.conf and zapata.conf there's plenty or info on the web to setup it. Hope you get the idea. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] Cdr error? Help!

the trick. And PLEASE do not make more threads of this, is you are not satisfied with the answers because of your lack of understanding, at least reply on the same thread, giving more details about your setup and making an effort to understand what people is trying to explain. -- Ing. Miguel Molina

Re: [asterisk-users] queue_log in mysql and file

another files) to make it do both types of logging, text file and realtime engine. A backend modular system similar to the CDR handlers actually present in asterisk, would be awesome to handle the queue logs too. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] queue_log in mysql and file

Lenz Emilitri escribió: You should log to a file and use a piece of code like our qloaderd to do the DB update. l. Could you share such piece of code? Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] mysql error (err 2002)

harry R escribió: mysql -uasterisk -pasterisk asteriskdb When I do that in a linux terminal it works. But I always have this err 2002. Greeting missing. Elaborate missing. Err 0x1b5a9f4c You're not talking to machines here. :-) -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone

Re: [asterisk-users] Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-)

that it supports. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] Asterisk Autodialer

and initial setup are far away difficult to understand (IMO). More than that, you won't find anything else on the scope of Open Source dialers for asterisk (AACC - Hanashi Dialer is in a very alpha stage). Anything else is closed and/or commercial. Cheers, -- Ing. Miguel Molina Grupo de Tecnología

Re: [asterisk-users] queue issue

asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, Maybe maxlen = 1? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] Asterisk PBX causes mysql to take more CPU time

version into production. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http

Re: [asterisk-users] T1 TE121 cable connected to a TN2464BP AVAYA card

. RJ48C/RJ48C crossover cable specifications That'll do. Cheers, -- Ing. Miguel Molina Grupo de Tecnologa Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix

Re: [asterisk-users] invalid extension

it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] Using asterisk as the recording server

and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth

Re: [asterisk-users] Using asterisk as the recording server

easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Erik de

Re: [asterisk-users] Realtime static with Asterisk 1.6.1.6

Carlos Chavez escribió: I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. Just curious, is there any specific reason for you to upgrade from the latest 1.6.0.14 to 1.6.1? Cheers, -- Ing. Miguel Molina Grupo de

Re: [asterisk-users] Asterisk CLI commands not running !!!!!

, googling error messages, dig into the bugtracker looking for reported issues, and don't give up, is better to have or achieve a stable version and maybe help to improve it reporting a new bug than just going with a lazy solution IMHO. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium

Re: [asterisk-users] Parser for Asterisk Queue Logs

, This should help: http://www.voip-info.org/wiki/view/Asterisk+log+queue_log At the end you will find the meaning of every field. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] [asterisk-dev] MeetMe in Macro

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Anahi Ludueña

Re: [asterisk-users] How to list ongoing calls from dialplan

to will tell you if it's bridged or not, and to what channel. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix

Re: [asterisk-users] MeetMe in Macro

) exten = s,n,Noop(I returned!) exten = s,n,Hangup [mysub] exten = s,1,Noop(So I'm at a subroutine) exten = s,n,Noop(I need to do special steps) exten = s,n,Playback(tt-monkeys) exten = s,n,Return() Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Anahi Ludueña escribió

Re: [asterisk-users] MeetMe in Macro

is ${var1}) exten = 8135551212,n,Noop(var2 is ${var2}) exten = 8135551212,n,Noop(var3 is ${var3}) ... and so on... with no need to call Macro() or Gosub(). Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Anahi Ludueña escribió: Thanks, I asked you to execute the GoSub from

Re: [asterisk-users] Music on Hold

for example. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] Voice Playback cutting first word or so of audio file

() Or answer(1000). Cool, didn't know about that one. One less line of code in the dialplan. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009

Re: [asterisk-users] Voicemail Crash - ODBC Realtime

Ekelund, Bryan escribió: Upon further review, it is not dumping out, just restarting on its own with the same error. No .dmp in /tmp Check that you are running asterisk with the -g option. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

[asterisk-users] About bug 13115

-10 days... Thanks for any pointers or help. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

Re: [asterisk-users] Asterisk 1.6 Transfer issue[Edited]

autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = call-limit = 20 member = SIP/100 member = SIP/101 member = SIP/102 Please help , I m in a total mess ...Thanks Sriram Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] GoTo IF

. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list

Re: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4

, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing

Re: [asterisk-users] multiple call

language. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net

Re: [asterisk-users] AMI 1.0 - 1.1 with originate.

/manager_1_1.txt Hope it solves your issue. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] how to limit the calls leaving a queue?

it was. That's all I can think of, without a hard work of messing with app_queue.c source code. Hope you get the idea. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] queues autopause

/asterisk-users Hi, Some months ago there was a discussion about this, with a simple solution involving minimal changes to the source (1 line of code). Search the archives of this list and you will find the answer. BTW, what version of asterisk are you using? Cheers, -- Ing. Miguel Molina Grupo

[asterisk-users] Cancel attended transfer

with *, it would end up in a blind transfer for the second agent who takes the transferred call. Is there a feature that can be configured to do that? I'm using asterisk 1.4.22 on a production server. TIA, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

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