Re: [asterisk-users] WSS over Asterisk

El 11/06/2014 1:52 p. m., Matthew Jordan escribió: On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington > wrote: Chrome 35 broke all of this you need to be using DTLS now I believe. I had working secure web sockets with asterisk 12.2.x and chrome

Re: [asterisk-users] unable to transfer ???

El 27/04/2014 8:39 p. m., Sean Darcy escribió: On 11.9.0: -- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz: --> requested format = speex, --> requested prefs = (), --> actual format = ulaw, --> host prefs = (silk16|ulaw|gsm|g722), --

Re: [asterisk-users] Queue callers with Callback option without lose their place

Known as Virtual Hold, you'll have to program inside asterisk to achieve that. El 31/05/12 10:48, equis software escribió: Is there any option in Asterisk distribution of this? Thanks. -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] No more CDR record for simple Hangup?

El 08/08/11 11:46, J Gao escribió: On 11-08-06 10:06 AM, Miguel Molina wrote: El 05/08/11 13:20, J Gao escribió: I am using the new 1.8.5 and I just found out that Asterisk won't record the call if the call just hangup. I did a test like this: exten => 1009, 1, Hangup() Then I cal

Re: [asterisk-users] No more CDR record for simple Hangup?

El 05/08/11 13:20, J Gao escribió: I am using the new 1.8.5 and I just found out that Asterisk won't record the call if the call just hangup. I did a test like this: exten => 1009, 1, Hangup() Then I called 1009: == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 -- Executing [1009@

Re: [asterisk-users] What is the use for the agent password if login via exten

El 18/07/11 18:03, bilal ghayyad escribió: Dears; If I need to login using as agent using the AddQueueMember(team,) then what to be the second paramter? How to be written? For example, if the agent id is 8000 then it will be: AddQueueMember(CustomerSupport,Agent/8000) or something else?

Re: [asterisk-users] How to create a module

El 08/07/11 12:50, Steve Edwards escribió: *) You can execute hundreds of AGIs written in C in the time it takes to load the Perl interpreter and parse your script. Just curious... have you timed this to demonstrate? --

Re: [asterisk-users] Access Voicemail Asterisk 1.8 FreeBSD 8.2

Hi, The VoiceMail() application is to leave messages in mailboxes, to enable users to access their voicemails, check the application VoiceMailMain(). core show application VoiceMailMain Regards, El 09/06/11 17:57, motty.cruz escribió: Hello, I'm new to this list. I'm trying to configure my A

Re: [asterisk-users] Do not disturbe

') -- Executing [...@a2billing:5] Hangup("SIP/2015-0187", "") in new stack == Spawn extension (a2billing, *11, 5) exited non-zero on 'SIP/2015-0187' Therefore, the

Re: [asterisk-users] Asterisk on smartphone?

and far better than a cellphone! ;-) -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductor

Re: [asterisk-users] Context issue

en of course I can't use that account for outbound calls..) Adrian If you use host = dynamic, I think the device must register with Asterisk for incoming calls go to the right context. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Ph

Re: [asterisk-users] MFC/R2 detected as ISDN PRI

cr2_logging=all > channel => 1-15 > channel => 17-31 > > Does they make sense to you? I'm kind of messed up after review them > hundreds of times in 48 hours. > > Cheers, > > Martin > > There you are... your system.conf has: span = 1,1,0,ccs,hdb3 MFC/

Re: [asterisk-users] Get the Uniqueid of Action Originate in the AMI

n the response of the action. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory we

[asterisk-users] MoH streamers for asterisk

tell me what streamers can be used with asterisk to stream say, gsm or wav files for MoH? I'd appreciate this info, we usually use 'files' mode and changing that could lower the load on asterisk. Thanks, -- Ing. Miguel Molina Grupo

Re: [asterisk-users] Migration from 1.2 to 1.8 in production

for what I think the asterisk team wants, that is, to focus on only one well supported version instead of having to support several parallel branches which mean more work and "cross-fixing" between them. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Milleniu

Re: [asterisk-users] Re : thousands Hangup per second /saturation of bandwidth

You didn't attach some debug output that shows some work, and you didn't even tell us what asterisk version are you using, which scenario is on, etc. Don't expect people to run and answer right away with an inmediate solution to this. -- Ing. Miguel Molina Grupo de Tecnología

Re: [asterisk-users] Licensing of Default MOH

s/by-sa/3.0/ > > This explicitly allows "public performance" and such. I suspect it > should be good for you. > > (Did I mention I'm not a lawyer?) > I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that came with asterisk. I wonder why the c

Re: [asterisk-users] Difference

ddons for 1.4. The good old (now unsupported) 1.2 works for many people, ask Steve. So it's up to you. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center El 06/10/10 11:04, Zeeshan Zakaria escribió: For a production environment, 1.4 is the most stable, and it has

Re: [asterisk-users] SIP flood attacK

El 04/10/10 10:35, khalid touati escribió: actually same thing happened to us a year ago (under asterisk 1.2) we solved the same day discovered by putting both: allowguest=no alwaysauthreject = yes Thanks for the tip! Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 2

hat's what I get for looking at the Dial command (timeout in > seconds) for most of the day then trying Originate (timeout in > milliseconds). > > You're right, I've had this same issue using the Originate command. This kind of timeout settings should be unified to av

Re: [asterisk-users] How to avoid interruptions with DIGIUM

http://wiki.sangoma.com/sangoma-hardware#aft_firmware Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

Re: [asterisk-users] Solving the CDR mess of attended transfers

, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] openvz

ok? > Outlook Express is a total PITA. Should I recommend you to use Mozilla Thunderbird... Sorry for the offtopic. And I agree, you should have no problems with asterisk using it inside an openVZ VPS. -- Ing. Miguel Molin

Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?

or 10 agents as noted. Go to your Avaya daddy... -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] IAX2 and other modules load error on 1.8.0-beta3

efined symbol: ast_aes_set_decrypt_key [Aug 23 08:31:54] WARNING[3883]: loader.c:819 load_resource: Module 'func_aes.so' could not be loaded. What are the requirements for these modules? Or is this an issue that needs to be reported on the bugtracker? Have a nice day. Regards,

Re: [asterisk-users] WaitExten() always times out

, you said that you were going to receive calls from the PSTN, but are you testing from a SIP endpoint? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by h

Re: [asterisk-users] WaitExten() always times out

g the CLI command "core set debug X", with X on 1 or 2, and setting it off when you want. If your call is received from the PSTN, asterisk will detect the inband DTMF tones in the audio signal. The rfc2833 configurations are only for VoIP endpoints. Good luck in your debugging

Re: [asterisk-users] Recording the conversation with MixMonitor() ends when the call is transfered

Hi, Never tried it, but you can take a look to the AUDIOHOOK_INHERIT function that allows MixMonitor to continue the recording in the same file after the transfer. http://www.voip-info.org/wiki/view/Asterisk+func+AUDIOHOOK_INHERIT Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium

Re: [asterisk-users] WaitExten() always times out

Hi, Are you sure asterisk is receiving and processing DMTF correctly? Are you using rfc2833, SIP INFO or inband DMTF? What is your asterisk version? I use WaitExten(5) all the time, no matter if they are single-digit or multiple-digit extensions. Regards, -- Ing. Miguel Molina Grupo de

Re: [asterisk-users] Codec Conversion

El 09/08/10 05:30, michel freiha escribió: Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina mailto:mmol...@millenium.com.co>> wrote: El 05/08/10 14:50, Tim Nelson escribió: - "mi

Re: [asterisk-users] Codec Conversion

t tends to have a > 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less > should not be using LPC10. > > -Jeff > > OK, on years I have working with asterisk I never have used, tested or even heard that old codec. I was just quoting the funny

Re: [asterisk-users] Codec Conversion

This just made me remember some comment on the iax.conf sample file... disallow=lpc10 ; Icky sound quality... Mr. Roboto. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwid

Re: [asterisk-users] Problem with Sangoma card...

ctor de Tecnología > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > My guess is that you need to upgrade the firmware of the Sangoma card. We had problems with a Dell server and a Sangoma card, solv

Re: [asterisk-users] How to transfer a call to operator using FAGI asterisk

ere are one line commands to achieve this. Finally, if you don't have control of your AGI and you need to make the transfer outside the AGI, simply do a Goto after the AGI to transfer the call where you need. Even asterisk itself gives you help: *CLI> agi show set context *CLI>

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

VPS can access it and then you could receive PRI calls or something directly into a VPS, or better, split channel groups between VPS. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and

Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

.3.1.el5.028stab069.6]# pwd /lib/modules/2.6.18-194.3.1.el5.028stab069.6 [r...@virtual1_ast1 2.6.18-194.3.1.el5.028stab069.6]# ls -lh total 44K lrwxrwxrwx 1 root root 57 jun 30 18:07 build -> /usr/src/kernels/2.6.18-194.3.1.el5.028stab069.6ent-i686/ I had to create the folder with the kernel

Re: [asterisk-users] Asterisk 1.6 (and 1.4) DTMF problems using RFC2833

F too but the features are not triggered... Anyone else has run into this? Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] "Hidden" memory leak

El 24/06/10 05:05, Tzafrir Cohen escribió: > On Wed, Jun 23, 2010 at 04:27:20PM -0500, Tilghman Lesher wrote: > >> On Wednesday 23 June 2010 15:45:05 Miguel Molina wrote: >> >>> Hi all, >>> >>> Anyone know why this happens? >>> >

[asterisk-users] "Hidden" memory leak

ons in file 'xmldoc.c' 99512 bytes in 239 allocations in file 'app_queue.c' 23080 bytes ( 23080 cache) in 73 allocations in file 'frame.c' 3308 bytes in 139 allocations in file 'pbx_config.c' 379223 bytes in 1940 al

Re: [asterisk-users] when to use e1/t1 card?

El 21/06/10 14:04, Necati Demir escribió: > This is a really rookie question: when should i use TE110P ISDN PRI Card? > > -- > Necati DEMİR > --- When you need to... -- _ -- Bandwidth and Colo

Re: [asterisk-users] DAHDI volume

r if there is any background noise at all. If > this is documented, point me to where and I'll gladly do my reading. > > Thanks, > --Greg > > > > Hi, Look for rxgain and txgain config options in http://www.voip-info.org/wiki/view/chan_dahdi.conf Regards, -- Ing. Mig

Re: [asterisk-users] Need fax solution for 1.4.xx

't mention native softfax termination. > > My best answer is 'I don't know', but the absence of mention seems > significant. > > Take a look at http://sourceforge.net/projects/agx-ast-addons/ There is fax support for 1.4 inside that modules. Cheers, -- Ing. Mig

Re: [asterisk-users] Help with FastAGI server in Windows

have experience with AGI but fastagi dont This is the third thread you have created for this. You're boring me now. S Why don't to try it in Java? Use the asterisk-java library (http://www.asterisk-java.org/) and you can run the FastAGI server in any OS. Cheers, -- Ing. Mig

Re: [asterisk-users] Is svn.asterisk.org down ?

nation Host Unreachable -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] Dynamic agent showing as "Invalid"

e.so module for it to recheck the local channel member definitions. Hope it helps. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

Re: [asterisk-users] This is a test, hijack this

Gergo Csibra escribió: > Hello Asterisk, > > This is only a test, because I can't start new thread in this list... > > If you can send an email, you can start a new thread on this list. What's the point of all this? -- Ing. Miguel Molina Grupo de Tecnolog

Re: [asterisk-users] [solved] Installing cdr_pgsql on asterisk 1.6.0.26

Miguel Molina escribió: Hi folks, I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the ./configure, it complains about the function PQescapeStringConn not existing in -lpq, so when I do a make menuconfig, I can't select the cdr_pgsql module. I am using CentOS 5.4

[asterisk-users] Installing cdr_pgsql on asterisk 1.6.0.26

y for 8.4 version. Some of the installed packages are postgresql-libs, compat-postgresql-libs, postgres, postgresql-contrib. Any help would be very appreciated. Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil:

Re: [asterisk-users] Using asterisk as avaya definity recording server

that you put the subject in all caps. He normally gets upset with everyone that does this on the subject or in the body. I've corrected the caps in the subject to avoid further upsetting. Cheers, -- Ing. Miguel Molina G

Re: [asterisk-users] AMD: HANGUP

k1.4/logger_8h.html#6ff63e8955665c4a58b1598f2b07c51a>, "Got hangup\n"); 00211 strcpy(amdStatus, "HANGUP"); 00212 break; 00213 } So basically check that the channel is not being hungup during application execution. Regards, -- Ing. Miguel Mol

Re: [asterisk-users] subject: 1.4 vs 1.6

eleased branch, no matter if it's LTS on the new release schema, will need time and a large user base that adopts it to report bugs and help stabilize it. I would not underestimate the actual 1.6.X branches. Just IMHO, any opinions welcome. Cheers, -- Ing. Miguel Moli

Re: [asterisk-users] string length in dialplan

0's at the beginning until it > is 10 digits in length. > My PRI provider needs it set to 10 digits always. > > Thanks, > > Jerry > > *CLI> core show function LEN -= Info about function 'LEN' =- [Syntax] LEN() [Synopsis] Returns the length of the argume

Re: [asterisk-users] Product offerings from DIDforSale

Neha Khandelwal escribió: /_Our Product offerings: _/ //Use the asterisk-biz list instead to advertise your asterisk-related products. Regards. // -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] chan_local and Originate

the latest 1.4.30-rc2 a try. 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 u

Re: [asterisk-users] Important security alert: update your dialplans now!

. Ok, if I get it the simplest workaround would be changing this: exten => _X.,1,Dial(SIP/${EXTEN}) To this: exten => _X.,1,Dial(SIP/${FILTER(0123456789,${EXTEN})}) If you're intended to receive only numbers from the dialstring, right? See http://www.voip-info.org/wiki/view/Asteri

Re: [asterisk-users] Asterisk and Faxing

ort a maximum of 9600bps. Cheers and Merry Christmas, -- 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 op

Re: [asterisk-users] Connect Two Asterisk's using isdn Cards

is: http://downloads.oreilly.com/books/9780596510480.pdf The only thing is to configure the appropriate extensions to Dial() through the DAHDI trunk between servers. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocatio

Re: [asterisk-users] Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')

oppy voice. Earlier I > integrated server 3 to server 1 and it was a smooth run. > > Any idea what could be the possible reasons! > > /ag Please provide the asterisk version and g729 codec that is installed on each server, so people can have a clue of what's happening. Maybe

Re: [asterisk-users] Connect Two Asterisk's using isdn Cards

an see the rest of the configurations on this link: http://www.voip-info.org/wiki/view/chan_dahdi.conf Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Cancel attended transfer

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 Hi Danny, Seems reasonable, but I can't use that because I'm using permanently logged in (AgentLogin) type agents, and that would requi

[asterisk-users] Codecs with MixMonitor (,a) option

Hi all, Another simple question: does it make sense to use the append option in MixMonitor (,a) when the codec is gsm? Or it works only when the codec is an uncompressed one like ulaw, alaw or slin? Thanks, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center

Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?

ous), which is clearly faster. However, the call file solution Danny proposed would work pretty well in your case. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] How to dial multiple extensions at once like in aring group and put them in conference?

Zeeshan Zakaria escribió: > Hi Danny, > > This is exactly what I am doing, but it takes a few seconds before all > the extensions are ringing. The loop takes its time. > Are you originating the calls asynchronously? -- Ing. Miguel Molina Grupo de Tecnología Milleni

[asterisk-users] Cancel attended transfer

*, 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 C

Re: [asterisk-users] queues autopause

rch the archives of this list and you will find the answer. BTW, what version of asterisk are you using? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

out 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-digital.com -- asterisk-users maili

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

AMI 1.0 to 1.1 take a look of these files on your asterisk source code: UPGRADE-1.6.txt doc/manager_1_1.txt Hope it solves your issue. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provi

Re: [asterisk-users] multiple call

ood one made on your favorite programming 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

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

ce, maybe someone else can enlighten us. 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:

Re: [asterisk-users] GoTo IF

unc+exists 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 asterisk-

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

time = autofill = yes 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 M

[asterisk-users] About bug 13115

ery 7-10 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 Reg

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

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

nswer() 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-digi

Re: [asterisk-users] Music on Hold

les. Try renaming to chopin_op40-1 and chopin_op40-2 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 -

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

Re: [asterisk-users] MeetMe in Macro

ub(mysub,s,1) 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 Mil

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

, the last field "Brigded 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 --

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 Lu

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

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

tom of it, try upgrading, 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

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 Gr

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

relies on the native recording capabilities which makes things really >> 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 minim

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

hings really 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

Re: [asterisk-users] invalid extension

(sorry, extension doesnt exist) ; all 1 digits exten exten => _X,n,Hangup exten => _XX.,1,Goto(s,1) ; accept exten LEN >1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Mill

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

cable Table C-5. RJ48C/RJ48C crossover cable specifications That'll do. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 -

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

y before putting the new 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, Arizon

Re: [asterisk-users] queue issue

13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Hi, Maybe maxlen = 1? Cheers, --

Re: [asterisk-users] Asterisk Autodialer

internals 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

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

ike the Playback() application, asterisk should look for the filename in any compatible format that it supports. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www

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 Tecnol

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 ___

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

e one or another. You would need to modify the source code of logger.c (and maybe 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. Cheer

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

do 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 Mol

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

exten => _3XXX,1,Dial(IAX2/ast1) exten => _3XXX,n,Hangup() exten => _3XXX,1,Dial(IAX2/ast2) exten => _3XXX,n,Hangup() That way you get an incoming ISDN PRI to SIP or IAX2 gateway. Modify the dialplan patterns according to your needs. For your PRI zaptel.conf and zapata.conf there'

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

ion you can ask me. > Are you storing the CDR in a database? What 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. M

Re: [asterisk-users] Inbound Call coding

ething. With all 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

Re: [asterisk-users] Server linux requirements

used. 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 ma

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

looks like 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.

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

(tt-weasels) > exten => 997,3,Hangup() > > exten => 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 > T

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