similar to: Asterisk and Media gateway controller

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

Disclaimer: Yes, I know faxing over G711 is unreliable. :-) We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're running Sipura SPA-2002's as ATA's and faxing within our own voice network is working. If we try and fax out to the world however, we're running into a problem. When the call connects and the modem tones begin to negotiate, our SIP/PSTN

Hi, I'd like to realize the following setup traditional phones --> pbx (non ip ) --> asterisk --- ISDN network is it possible to set up an asterisk server in such way that it acts as gateway between a traditionnal pbx and an isdn network ( I'm in France ). this to let people continue with their existing pbx + phones and enjoy voip features ? outgoing calls: if yes,do I need

Hola, Shiny utiliza la función selectInput en ui.R (donde se declaran los métodos y las etiquetas que se muestran en pantalla) y la variable input$nombre_usado_en_ui para tomar el valor elegido en el server.R Puedes consultar el tutorial de shiny en: http://rstudio.github.io/shiny/tutorial/#shiny-text Por ejemplo (copiado de ahí) en ui.R sidebarPanel( selectInput("dataset",

Hello on Behalf Of idont know, Sangoma has a Media Gateway solution via SS7. They I believe are the only ones capable of connecting Asterisk via SS7. You may want to check them out. Heidi -----Original Message----- From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of idont know Sent: April 6, 2006 10:29 AM To: asterisk-biz@lists.digium.com

Is anyone using a media gateway with a DS3 for TDM/PSTN access? I am trying to determine the best way to scale Asterisk beyond 4 PRI's. What about the TNT Max? Can it be configured with a DS3 PSTN interface, the appropriate software and hardware, and Ethernet ports to support SIP-PSTN/DS3 media gateway services for multiple * boxes? What about TNT - SER - * ? Is there anyone here with real

Hello. I'm a little surprised with the set of VoIP protocols asterisk supports. But I can't find MEGACO. Is there any known reason or problem with Asterisk not to support MEGACO? Thanks, Arsen. __________________________________ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail

Hello ALL! 1. eviroment: 1) PBX:Asterisk@HOME(Asterisk 1.2.7.1) 2)Media GateWay:Grandstream HT 488(register to pbx using SIP account,such as 9001),connect to PSTN(phone no is 82820088) using its FXO. 3)Queue:set up a queue named [myqueue],dial 2020 can call queue 2.my question: when customers need help ,they call 82820088, and then call 2020 to enter the queue . From now on ,the

Hola, no suelo usar dropbox para dejar ficheros de datos pero no he tenido ningún problema para descargar un dataset (respondiendo a una consulta en un foro linkedin). Por ejemplo: file_url = "https://www.dropbox.com/s/1j0pdi02asc3g20/BigDataE1.csv?dl=1" kk <- read.table(file_url, header = TRUE, sep = ';', quote = "\\")

Sir, Here is the working Audiocodes MP-11X FXO configurations to work with Asterisk. ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 905371 ;Slot Number: 1 ;Software Version: 5.00A.024 ;DSP Software Version: 204IM => 209.13 ;Board IP Address: 192.168.0.195 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to

Hola: Por si no os ha llegado por otras vías, os informo que se están gestando las próximas (¡las 4as!) Jornadas de Usuarios de R en Castellano. Ved el mensaje más abajo de Juan Ramón González. En breve, los que queráis echar una mano, hacer sugerencias, etc. , podéis hacerlo en el foro que se ha creado en la web de la comunidad (hay gente que no está - aún - en esta lista, etc):

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 14th and 17th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral Symposium is to provide graduate

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 14th and 17th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral Symposium is to provide graduate

More top posting goodness... Please post your updated dialplan. After making the change, did you reload/restart Asterisk so the changes would take effect? --Tim ----- "Mohit Saxena" <MohitS at starcomms.com> wrote: > Still no luck.... > > Br, > Mohit > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral

------------------------------------------------------------------ Doctoral Symposium of CISTI'2017 12th Iberian Conference on Information Systems and Technologies 21th and 24th of June 2017, ISCTE-IUL, Lisboa, Portugal http://www.cisti.eu/ --------------------------------------------------------------------------- The purpose of CISTI'2017?s Doctoral