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If there any guide / howto yo Bulls a vernos live cd? -- Imagination is more important than knowledge

On 10/23/20 4:50 PM, Leonardo Cuyar Morales wrote: > Hi list, sorry about my autocomplete. I asked about any documentation on > build a centos livecd with my custome utities and applications. > > On Saturday, October 17, 2020, Leonardo Cuyar Morales <leo.cuyar at gmail.com> > wrote: >> If there any guide / howto yo Bulls a vernos live cd? >> >> -- >>

Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y

Hi list, sorry about my autocomplete. I asked about any documentation on build a centos livecd with my custome utities and applications. On Saturday, October 17, 2020, Leonardo Cuyar Morales <leo.cuyar at gmail.com> wrote: > If there any guide / howto yo Bulls a vernos live cd? > > -- > Imagination is more important than knowledge > -- Imagination is more important than

Does any know what happens with svn repository on svn.digium.com ? -- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y de pronto estar?s haciendo lo

Dear asterisk users, We want setup TE121 digium board: Model: Digium TE121: VoiceBus technology allows the TE121 to use an industry standard bus-mastering PCI Express interface. http://www.digium.com/en/products/digital/te121.php My platform Server: HP Proliant 150 G5 OS: UBUNTU x86_64 GNU/Linux Asterisk: 1.4.21.2 zaptel: SVN-branch-1.4-r4662M When we enable zaptel driver for TE121, the

Folks, I've seen a few reports that people have had problems with hang up detection on UK cable phone lines. I have a TDM400P with two FXO ports, one connected to my BT line and the other connected to my Telewest/Virgin Media cable line. If I ring the BT line and then clear down, Asterisk detects this and acts accordingly. If I ring the Telewest line, the clear down is not detected, hence

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Hola, Tengo la base de datos de un censo agropecuario el cual está en formato de SPSS, cuando cuando ejecuto el comando para introducirlo a R, despues de permanecer lento R, aparece esto: Error: cannot allocate vector of size 1.6 Mb Hasta aquí llega y no introduce ninguna observación. Mi pregunta es si existe alguna solución, talvez aumentar la memoria de R?? Imagino que como ustedes han

Hi All I'm having some serious kernel panic while using digium cards. It may be related to IRQ shared. Can this cause a lot of drop call and bad voice quality ? Do you guys know if there is a way I can assign one IRQ for each digium card ? Thanks a lot. Here is the output of /var/log/syslog kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20. kernel: [

Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot

Any hardware that can do 25-50-100 fxs ports trunked to sip ? Example one end a cat5 other end 50 RJ11's jacks.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090406/9514bc47/attachment.htm

Anyone have a cookbook on configuring sendmail to work with Asterisk? Or,a few config examples.

Is there a way in the dialplan to figure out which agent in a ring all queue answered a line? I'd like to take specific action based on the agent upon hangup. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue & McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hi, Does anyone have a suggestion how I can analyze the concurrent usage of ISDN channels? A client complains about their clients sometimes getting a busy tone when trying to call them. I suspect they just need to add an additional ISDN2 line but I need some conclusive information that they are indeed maxing out their ISDN channels. Thanks, Patrick

Dear All. I have the following table that I want to analyze using multinom function freq segments sample 4271 Seg1 tumour 4311 Seg2 tumour 3515 Seg1 normal 3561 Seg2 normal I want to compare model with both factors to the one where only sample is present. model1=multinom(freq~segments+sample,data=table) model2=multinom(freq~ sample,data=table)

Need some assistance, I have a firewall I am trying to reinstall the os, on, and I have no idea how to do it, they sent me syslinux, but I have no idea what kernel to use, or really how to install it using syslinux. Please, any help will be appreciated. Scott Latta Network Administrator Techsupport Supervisor Nexity Bank A+, NET+, MCP 205-298-6482

People, I have an Asterisk SIP server and I have to connect my voip users to the PSTN via a E1 trunk. I have two ways to do this: 1) Connect a E1 interface to my Asterisk (is it possible ???) and then connect my Asterisk to the PSTN 2) Buy a commercial IP PBX with a E1 interface, generating the SIP users again, and connect the IP PBX to the PSTN What do you recommend to me based on your