Re: [asterisk-users] Problem with extensions in IVR and queues

Hi, we've just been able to find the problem. Apparently it was related to the softphone. We've installed another one and the call is performed ok. Thanks! Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 19:59:14 + Subject: Re

Re: [asterisk-users] Dial options not working

options not working Anahi, What kind of line do you have ? POTS, PRI, SIP ? It seems like the DTMF is not coming in correctly or you have some bad settings on your end. - Original Message - From: Anahi Ludueña To: asterisk-users@lists.digium.com Sent

Re: [asterisk-users] Dial options not working

Hi, yes, I've just tried to use the dtmf mode inband, but it doesn't work with landline phones or cell phones... Thanks, Anahi Ludueña Date: Wed, 30 Jun 2010 12:56:59 +0100 From: kwat...@geniusgroupltd.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dial options

[asterisk-users] Problem with extensions in IVR and queues

else to set? Thanks, Anahi Ludueña _ ¿Un navegador seguro buscando estás? ¡Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com

Re: [asterisk-users] Problem with extensions in IVR and queues

Thanks Danny, but I don't know what I should do to fix it... Could you help me? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010 10:33:31 -0500 Subject: Re: [asterisk-users] Problem with extensions in IVR and queues

Re: [asterisk-users] Problem with extensions in IVR and queues

[...@ivr-3:13] WaitExten(SIP/9050-001185aa, |) in new stack == Spawn extension (ivr-3, s, 13) exited non-zero on 'SIP/9050-001185aa' -- Executing [...@ivr-3:1] Hangup(SIP/9050-001185aa, ) in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9050-001185aa' Anahi Ludueña

Re: [asterisk-users] Problem with extensions in IVR and queues

Ups, sorry, that CLI output is related to my other problem (the options of IVR doesn't responde when the call is from landline or cell phone). I'll put the correct CLI output... Thanks, Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 30 Jun 2010

[asterisk-users] Dial options not working

. When I press one option, it seems I do nothing... Please, could you help me? Thanks, Anahi Ludueña _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http

Re: [asterisk-users] Dial options not working

Thanks, but I don't have any *dahdi*.conf file here... (I check in /etc/asterisk) Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 16:54:01 -0500 Subject: Re: [asterisk-users] Dial options not working Check your DTMF

[asterisk-users] FW: Music on Hold problema

Please, I need help with this... Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 15:12:25 + Subject: Re: [asterisk-users] Music on Hold problema The list of /var/lib/asterisk/mohmp3 is: -rw-rw 4 asterisk asterisk

Re: [asterisk-users] FW: Music on Hold problema

the music on hold. Thanks, Anahi Ludueña Date: Wed, 23 Jun 2010 10:44:10 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Music on Hold problema The moh conf file seems good. It is the standard implementation and should have worked

Re: [asterisk-users] Music on Hold problema

Any ideas, please? Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 19:54:30 + Subject: Re: [asterisk-users] Music on Hold problema I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da

Re: [asterisk-users] Music on Hold problema

=files directory=/dev/null Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 18 Jun 2010 09:26:16 -0500 Subject: Re: [asterisk-users] Music on Hold problema Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf

[asterisk-users] Music on Hold problema

[20784] logger.c: -- Stopped music on hold on SIP/7PBX-08229d18 [Jun 17 19:20:22] DEBUG[20784] channel.c: Scheduling timer at 0 sample intervals Could you help me with this? Thanks, Anahi Ludueña

Re: [asterisk-users] Music on Hold problema

I have wav files in the /var/lib/asterisk/mohmp3... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 17 Jun 2010 14:36:00 -0500 Subject: Re: [asterisk-users] Music on Hold problema I see that moh is trying sln format, then ulaw

Re: [asterisk-users] Call ended after 31 seconds

Yes, I'm using XLite... Anahi Ludueña From: l...@virtutel.ca To: asterisk-users@lists.digium.com Date: Fri, 11 Jun 2010 20:05:39 -0400 Subject: Re: [asterisk-users] Call ended after 31 seconds You`re using Xlite/eyeBeam by any chance? Mike From: asterisk

[asterisk-users] Call ended after 31 seconds

: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3000-6d07' in macro 'dialout-trunk' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (from-internal, xxx, 5) exited non-zero on 'SIP/3000-6d07' Thanks, Anahi Ludueña

[asterisk-users] Press twice *

Hi people, I need to detect when the user presses twice *... In the dialplan I added the following, but it doesn't work. Could you help me with that? exten = **,1,. Anahi Ludueña

[asterisk-users] Dial an extension with follow me

) == Auto fallthrough, channel 'SIP/CALLUS-0b3f' status is 'CHANUNAVAIL' Thanks, Anahi Ludueña _ Aprende los trucos de Windows 7 con la gente que ya lo han probado Windows 7. http

[asterisk-users] Playback in h extension

Hi people, I'm trying to execute the PlayBack command in the h extension... but it is not played... is it possible to do that?Thanks, Anahi Anahi Ludueña _ Ahora Messenger en tu

[asterisk-users] AMI logs

Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludueña

[asterisk-users] Manager Logged off

Hi People, I don't know if my problem should be reported in this forum, but maybe somebody knows about it. I'm using the tool .NET WebService Studio to test the web service which is working with asterisk by AMI. It is working fine, the dialplan is executed correctly... the problem is when the

[asterisk-users] Voicemail after hangup

Hi people, just a question: Is it possible to execute Voicemail command in the h extension? (after hangup the channel). Because if I put it before it, it works right, but if I put it there, it doesn't... The log is: -- Executing [...@cont-mine:1] NoOp(SIP/3005-096736a8, End of cont-mine)

[asterisk-users] Hangup

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten = h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks,

Re: [asterisk-users] Hangup, SoftHangup

Thanks Phillipp!, it works! Anahi Ludueña Date: Tue, 10 Nov 2009 14:44:09 +0100 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup, SoftHangup Anahi Ludueña schrieb: is it possible to hangup a channel from another channel

[asterisk-users] Voicemail file

Hi all, When somebody leaves a message in the voicemailbox, is there a way to know the file name of it? I need to return the voicemail file name in the deadagi command. Thanks, Anahi _

Re: [asterisk-users] Voicemail file

Yes, but is there a way to know the filename of that message? For example: msg0029.wav? I know where it is saved, but if I want to return it, I need to find the last one... and it is not recommended in my opinion... Thanks, Anahi Ludueña From: da...@debsinc.com To: asterisk-users

Re: [asterisk-users] Voicemail file

Thanks people, I've already found the way... The variable ${VM_MESSAGEFILE} contains what I need... Bye, Anahi Ludueña From: a_ludu...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 30 Oct 2009 15:18:25 + Subject: Re: [asterisk-users] Voicemail file Yes

Re: [asterisk-users] R: R: R: CDR(billsec)

Thanks Matt! It works now! Bye... Anahi Ludueña Date: Fri, 30 Oct 2009 01:20:02 +1300 From: li...@venturevoip.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] R: R: R: CDR(billsec) On 30/10/09 1:10 AM, Alexandru Oniciuc wrote: Thank you! My bad,the CDR

[asterisk-users] CDR(billsec)

? In that case, how can I get the call duration in the h extension? Thanks, Anahi Ludueña _ Infórmate, mantente en contacto y encuéntralo todo, a la vez. Con la nueva Toolbar de MSN

Re: [asterisk-users] Dial a external number with extension

Thanks, I need to make a conference between 2 numbers, one of them is external and it has an extension. So, I need to dial the number and later enter the extension, how can I do that? _

[asterisk-users] Dial a external number with extension

Hi People, I need to dial an external number, when it is answered, I should digit the extension. How can I do that in the DialPlan? Thanks, Anahi Ludueña _ ¿Sabías que ahora puedes

[asterisk-users] Dialplan problem

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten = 2001,1,Answer exten = 2001,n,Dial(local/3005) exten = 2001,n,Hangup exten = 3005,1,Set(__RINGTIMER=10) exten = 3005,n,Macro(exten-vm,novm,3005) exten = 3005,n,Hangup When I

Re: [asterisk-users] Dialplan problem

Thanks, the answers helped me... I was thinking to execute a macro or another context which performs a DIAL command to a particular number. First I checked how it was working doing DIAL directly... that is the reason why I put that context. Thanks again... Anahi Ludueña Date: Fri, 9

[asterisk-users] OriginateResponse Event

of each one? Thanks in advance... Anahi Ludueña _ Nuevo Windows Live, un mundo lleno de posibilidades. Descúbrelo. http://www.microsoft.com/windows/windowslive

Re: [asterisk-users] OriginateResponse Event

Thanks Danny, but how can I get it from my web service? Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 5 Oct 2009 10:03:41 -0500 Subject: Re: [asterisk-users] OriginateResponse Event Each response set has a uniqueid field

[asterisk-users] Followme

, I'm using the Originate. It needs a channel as an argument, so the context can be executed; but what channel should I pass there? (the phone numbers are in the Variable argument) Thanks in advance. Anahi Ludueña

Re: [asterisk-users] Followme

-users] Followme Local/1 will run the context without tying up resources. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, October 02, 2009 8:20 AM To: asterisk-users@lists.digium.com

Re: [asterisk-users] Followme

Thanks, anyway the result is the same... Response: Error Message: Originate failed Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 10:01:05 -0500 Subject: Re: [asterisk-users] Followme Change the 1’s to s

Re: [asterisk-users] Followme

Maybe there is another problem. I changed the context like you said. Where is the local channel configured? or is it implicit? Sorry but I'm newbie with Asterisk... Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Fri, 2 Oct 2009 10:15:45 -0500 Subject: Re

[asterisk-users] How to finish a Meetme

, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive

Re: [asterisk-users] UpdateConfig

Thanks, It worked, it seems there was something wrong. The following is working now: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-00: Append Cat-00: default Var-00: 2000 Value-00: ,Jhon ActionID: 1234 Bye, Anahi Ludueña Date

[asterisk-users] UpdateConfig

the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y

Re: [asterisk-users] UpdateConfig

Thanks, the result was: Response: Success Anahi Ludueña From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Tue, 29 Sep 2009 15:16:52 -0500 Subject: Re: [asterisk-users] UpdateConfig Two questions: 1. do you need an ActionID line? 2. did you try

Re: [asterisk-users] MeetMe in Macro

Hi Juan, I didn't use the GoSub application, I put the name of the context in the Originate and the variables and their values in the Variable field. See http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate. Good luck! Anahi Ludueña From: jcard...@tpmex.com

[asterisk-users] MySQL cmd

Hi people, I'm trying to retrieve data from the database (server MySQL). I have the following dial plan: exten = s,1,Noop(Start) exten = s,n,MYSQL(Connect connid localhost user pass asteriskcdrdb) exten = s,n,Noop(Connid: ${connid}) ... The problem is that the 3º line is not showing the

Re: [asterisk-users] DeadAgi

Thanks guys, I'll take it into account!... Anahi Ludueña Date: Fri, 18 Sep 2009 10:13:12 +0100 From: i...@pack-net.co.uk To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DeadAgi Steve Edwards wrote: On Thu, 17 Sep 2009, Anahi Ludue?a wrote: Thanks

[asterisk-users] Audio Files

Hi people, What can I use to transfer the audio files to and from Asterisk? I was searching and I found the following commands: PUT SOUNDFILE and GET SOUNDFILE They are new commands of AGI, but is there another way to do that? Thanks, Anahi Ludueña

[asterisk-users] DeadAgi

== finconf.php|800|: Failed to execute '/var/lib/asterisk/agi-bin/finconf.php': No such file or directory But the file is there. The command ls -l returns: -rwxrwxrwx 1 root root 140 Sep 17 15:42 finconf.php Why does it return the error? Thanks, Anahi Ludueña

Re: [asterisk-users] DeadAgi

Thanks for the answers! The file didn't have the first line! #!/usr/bin/phpBye! Anahi Ludueña From: tles...@digium.com To: asterisk-users@lists.digium.com Date: Thu, 17 Sep 2009 15:59:21 -0500 Subject: Re: [asterisk-users] DeadAgi On Thursday 17 September 2009 15:06:28 Geraint

[asterisk-users] MeetMe in Macro

Thanks Miguel, It was my mistake. So, my question is: if I want to call the GoSub application from the Originate Action (using AMI), what I need to put in the context parameter? The GoSub will jump to a special context. Thanks, Date: Wed, 16 Sep 2009 09:34:31 -0500 From:

Re: [asterisk-users] MeetMe in Macro

told me that GoSub could replace the Macro, I thought it could be called from the Originate... Do you know if there is another way to pass some parameters to a context from the Originate? Thank you! Anahi Ludueña Date: Wed, 16 Sep 2009 10:27:26 -0500 From: mmol...@millenium.com.co

[asterisk-users] Meetme feature

Hi People, I want to do the following steps: - Create a meetme between 2 persons. - First, 1 person (user1) is entered into the meetme. - Second, user2 is entered into the meetme. User2 is the marked user and also he is able to exit the conference by pressing #. - If user2 exited by pressing