similar to: Strange beep when using VoiceMailMain application

Hi, we are experiencing a problem that is very strange, only on SOME calls, a locution jumps in to the RTP stream and both persons between the phones can hear it. It is looped and it does not stop till hang up. Do you have any clue about what could be happening ? Thank you ! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly with a CISCO mediaGW in order to send faxes to the PSTN using T.38. When Asterisk sends the initial INVITE containing the T.38 media offer in the SDP, the CISCO answers with a 488 Not Acceptable Media. Apparently, it looks like a configuration problem in the CISCO, but I have tested the CISCO with the Zoiper

Hello, We have been working with the ReceiveFax application for some weeks now in order to receive faxes in T.38 and it works fairly well, but there are some faxes that for some reason we are not able to receive correctly. The asterisk version we are using is 1.6.0.6 with spandsp-0.0.5pre4 and the asterisk machine is behind a CISCO mediaGW to be able to communicate with the PSTN. The SIP call

I'm trying to resolve a problem of Namespaces and probably have misunderstood something basic. In my quantreg package the function crq() has as its first argument a formula that typically contains something like this: Surv(y, ...) ~ foo, where Surv is a function from the survival package which quantreg suggests. When this syntax is encountered in ordinary usage Surv is resolved, I believe,

Hello, In my scenario, the asterisk machine is installed behind a CISCO mediaGW in order to be able communicate with the PSTN. Asterisk is configured to use T.38 to send and receive faxes. I'm trying to receive a fax from a fax machine located in the PSTN. Apparently everything goes well: the fax machine says the transmission was successfully completed, and the fax file is successfully

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Bueno, realmente no es necesaria que la serie esté centrada en este caso, ya que estoy sumando un ruído blanco Un saludo From: fjroar en hotmail.com To: caaperezan en gmail.com; r-help-es en r-project.org Date: Wed, 5 Nov 2014 13:00:49 +0000 Subject: Re: [R-es] Agregar ruido a una serie de tiempo Hola buenos d?as: Yo cuando he tenido que hacer estos trabajos, lo que hac?a era coger la serie

imagen1 MEDICI?N POR TELEMETR?A DE VOL?MENES DE AGUAS NACIONALES Aplicaci?n de la NMX-AA-179-SCFI-2018 ??nete a nuestro curso online de capacitaci?n sobre la medici?n por telemetr?a de vol?menes de aguas nacionales y aplicaci?n de la norma NMX-AA-179-SCFI-2018! Fecha: 23 de Julio Duraci?n: 6 horas Modalidad: Online En este curso, aprender?s las t?cnicas m?s avanzadas para medir vol?menes de

Es posible agregar ruido a una serie de tiempo Tengo series de tiempo que tienen un comportamiento funcional, quisiera agregar ruido para que parezcan señales mas reales. Normalmente las series de tiempo se suavizan a traves de filtros, es posible hacer el proceso inverso con algun paquete de R Gracias por la atención CARLOS ANDRES PEREZ [[alternative HTML version deleted]]

If I have a text with 7 words per line and I would like to put first and second word joined in a vector and the rest of words one per column in a matrix how can I do it? First 2 lines of my text file: "2008/12/31 12:23:31 numero 343.233.233 Rodeo Vaca Ruido" "2010/02/01 02:35:31 palabra 111.111.222 abejorro Rodeo Vaca" Results: Vector: 2008/12/31 12:23:31 2010/02/01 02:35:31

M?XICO NOM-001-SEMARNAT-2021 Comunicado del 30 de Enero 2023 Actualizaci?n de la NOM-001-SEMARNAT-2021 Aplicaci?n Regulatoria Obligatoria a partir del 11 de Marzo 2023 Fecha del evento 27 de Febrero 2023 Online v?a Internet Capacitaci?n Indispensable para Negocios y Empresas que realicen Descarga de Aguas Residuales Este curso tiene el objetivo principal de dar a conocer la aplicaci?n,

M?XICO NOM-001-SEMARNAT-2021 Comunicado del 27 de Enero 2023 Actualizaci?n de la NOM-001-SEMARNAT-2021 Aplicaci?n Regulatoria Obligatoria a partir del 11 de Marzo 2023 Fecha del evento 24 de Agosto 2023 Online v?a Internet Capacitaci?n Indispensable para Negocios y Empresas que realicen Descarga de Aguas Residuales Este curso tiene el objetivo principal de dar a conocer la aplicaci?n,

I have a extension 981 setup for entering VoiceMailMain: exten => 981,1,VoiceMailMain,([mailbox]@usvm) exten => 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready. *CLI> -- Executing VoiceMailMain("SIP/2504-ba66",

Hi everybody, I'm in a middle of a Asterisk learning period. I am at a very good point except I'm not able to use VoiceMailMain. This Is my simple dialplan regarding VoiceMail ;Number that the IP Phones dial to access voice mail exten => 22999,1,VoiceMailMain (s${CALLERIDNUM}) exten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until

Hi in asterisk 1.4, I need to cancel the password check and allow users enter in the mailbox without entering password. I tried this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,n,VoicemailMain(${Mailbox}@default,s) exten => 911119,n,Hangup and this: exten => 911119,1,Set(LANGUAGE()=es) exten => 911119,2,VoicemailMain(s) exten => 911119,n,Hangup But it does not work,

Hi All, The situation: When I dial into VoiceMailMain(@context), put in my VM # 1001 and Password 1001, no problem, but at the voicemail main audio prompt (Alison), when I ?press 3 for advanced options? then ?press 5 to leave a message? I put in a mailbox number 1002 within the same [context], but VoiceMailMain looks for the mailbox in the [default] context and will not recognize the mailbox I?m