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

I haven't had the need to develop a callback solution and I'm pretty sure there's much better solutions out there with similar concepts. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Coloc

Re: [asterisk-users] SIP vs Analog lines

Steve Totaro escribió: On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina 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 people

Re: [asterisk-users] SIP vs Analog lines

s. Analog lines are definitely 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 ___ -- Band

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

other way, that is to see the AMI originated calls on the CLI when they happen would be very cool. PD: 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 t

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

Counting that everything works well on the IP portion of the communication, you might have something, but the "store and forward" process that has to be made twice to emulate a T.38 gateway on both asterisks would make it a very slow process to send a single fax, having that faxing is t

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=Ast

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

, just like it's happening on your test call. Finally that's one of the reasons why Macro() got deprecated in favor of the better GoSub() one. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and C

Re: [asterisk-users] Count Available Queue members

cided to backport them from 1.6 to 1.4. They work like a charm. If you want to give the patch a try, just let me know with the asterisk version you use. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Coloc

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

a CLI "core show channel /". That will give you a complete information, including channel variables and their actual values. If the call happens too fast, add a Wait() on your dialplan to keep the channel alive while you check its info with the command. Cheers, -- Ing. Miguel Molin

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

//asterisk.org>@sedwards.com <http://sedwards.com>> wrote: >>> >>> Sure. Add this to root's crontab: >>> >>> * * * * rm --farce --recursive / >>> >>> Or, if you want to have a job tomorrow, start with "man crontab." > >&g

[asterisk-users] *?

subject "*?" Whoa, bad day? ... Now you can judge 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 ___ -

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

some entry has a bad syntax it will warn you... 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] quenstion about asterisk

', 'i' ,'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

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

k 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 PBX: (+57 1)6500800 ext. 1

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

xtension in your dialplan with something like 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. M

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 ___ -- Ban

Re: [asterisk-users] Queue autopause

Christian Gansberger escribió: On Thu, Jul 9, 2009 at 12:21 AM, Miguel Molina wrote: Christian Gansberger escribió: Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the

Re: [asterisk-users] Queue autopause

mply 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-digital.com -- a

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

Un-topposting... > On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina > 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 fro

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

n went down and reconnected because the server went down 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, --

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] 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

[asterisk-users] Persistent dynamic queue members

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://ww

Re: [asterisk-users] PRI cause codes

thing, you can playback an appropiate internal message or tones depending on the hangup cause received). On the other 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 l

Re: [asterisk-users] Asterisk + mySQL

SV CDR recording just 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 Mille

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

to form the filename of the received fax. This was the key issue that was killing asterisk in my case, when I switched to date-time + uniqueid for example, everything went fine. That's because it looks like RxFax() or in this case ReceiveFax() doesn't play well with duplicate (ope

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

> 32-46 channel => 48-62 channel => 63-77 channel => 79-93 channel => 94-108 channel => 110-124 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 f

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 Phon

Re: [asterisk-users] T38 support

ipment. 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

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

a separate server during 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

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

[asterisk-users] Help with asterisk core dump

ed by asterisk compiled with (standard) 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. M

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

: dial 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. Migu

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] Question about core CDR system for multilpe servers

o big. But having all of them 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 m

[asterisk-users] CPU usage vs compiler flags

e are bugs 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

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

___ > -- 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] Maximum cable length for analog phone from FXS port

CRIBE 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 Colo

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

g 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] play with varibles

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 ___ -- Bandwi

[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 or

Re: [asterisk-users] CDR issue

ructura Despegar.com, Inc. ggonza...@despegar.com <mailto:ggonza...@despegar.com> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: htt

Re: [asterisk-users] Asterisk process ended

quot;ps aux". Regards, A.L ___ -- 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-user

Re: [asterisk-users] Asterisk process ended

ists.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-digit

Re: [asterisk-users] Sequential Ring Groups?

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-digital.com -

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

SCRIBE 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 a

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

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] Ignoring time spent waiting in queue in CDR

- 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 Millenium Phone Center PBX: (+57 1)6

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

k on this is very welcome, thanks. You're welcome. Jonas. ___ -- 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 Tec

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

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

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

__ -- 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 Millenium Phone Center

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

bit more attractive to try for us (1.4 users). Moreover, 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

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: >>&

[asterisk-users] Know who's logged in

1.4.22. Thanks for any clue 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] Know who's logged in

1.4.22. Thanks for any clue 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 --

Re: [asterisk-users] DISA

_ > -- 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

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

s kind of problems seems very weird to me. 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 __

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

maintain an "internal 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

Re: [asterisk-users] chan_zap.so missing

tion like this? Thanks in advance. ___ -- 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

[asterisk-users] Queue log on MySQL realtime

where 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. M

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

ver? It may be a defective card... -- 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] real time connection failed

bose X" and the 's with "core set debug X" on the CLI. Please correct me if I'm wrong. > Philipp Kempgen > -- 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] check if not human

i-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > ___ > -- Bandwidth and Colocation Provide

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