[Asterisk-Users] * website needs a place for
Hello! This should be a list to come find support and not get jumped on! The * website should instruct where to find information better. Often times the first response to trying to learn something is to ASK a question. I too, first found the archive list tonight. I've been on this list reading since February. Better documentation is the key and since this is a product being developed daily keeping up with the documentation is difficult. It's the new people coming in which keep this idea alive as we, who have been around tell them. What do people see when they read list mail? I see PJ trying to help and John B. who BTW, is also a VOIP reseller, jumping on people who are not changing subject lines. Education and documentation is key to making a product succeed. Possibly a * web page re-design would better educate new people coming into this list so they conform to the lists standards. Also a reminder to those who know far more than I, You too started someplace and someone answered your questions and you learned. Please, lets be considerate of others. Possibly an automated daily message could be sent to the list reminding people to change the subject line or provide a link to the archives... Helping people succeed with * helps everyone who has an interest. Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A WORKING EXAMPLE
Hello! I've looked at the reference examples they are all for SIP. I have two X100p and a TDM400P. Can someone send me a working example so I can receive calls and make them. I'm stuck at first base. [I'm using standard phones - not SIP] Help please! Thanks, Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 1 and 2 -Martin?
Martin, Your statement below is somewhat confusing. Where do you find the choice of 1 or 2? This is the latest voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=gsm|wav49|wav ; Who the e-mail notification should appear to come from serveremail=asterisk ;[EMAIL PROTECTED] ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message ;maxmessage=180 ; Maximum length of greetings ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; ; Each mailbox is listed in the form mailbox=password,name,email,pager_email ; if the e-mail is specified, a message will be sent when a message is ; received, to the given mailbox. If pager is specified, a message will be sent there as well. ; [default] ;1234 = 4242,Example Mailbox,[EMAIL PROTECTED] ;4300 = 3456,Ben Rigas,[EMAIL PROTECTED] ;4310 = 5432,Sales,[EMAIL PROTECTED] ;4069 = 6522,Matt Brooks,[EMAIL PROTECTED] ;4110 = 3443,Rob Flynn,[EMAIL PROTECTED] At 01:24 PM 9/12/2003 -0500, you wrote: you can copy voicemail.conf.sample to be your voicemail.conf ... Martin On Fri, 12 Sep 2003, Olle E. Johansson wrote: Steven Critchfield wrote: On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote: While on the subject of Voicemail - what is the difference between voicemail() and voicmail2() ? From the application stand point there is little difference, but from the configuration stand point there is a fair amount of difference. Consult the sample configs to start you on your path to deciding what you want. Steven, Thank your for responding. I find only one config in the sample directory - voicemail.conf.sample and it looks the same as my voicemail.conf - should I look in another place? /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * -- FWD
Hello! There is much info using SIP for this connection type. I have a TDM400P with a regular phone connected to it. What string in extensions.conf would need to be added so I can call a FWD number as well as receive a call from FWD? I have seen some mention of using IAX. Is this necessary or can this connection be accomplished without enabling IAX? Thanks, Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What I need to make this work?
Hello! I just joined and am looking around for a place to start. Would this Asterisk Developers Kit be a good place to start? This is what I'd like to do. I am a wireless ISP. Several of my customers have offices in different communities that I serve. The local telephone company charges 8-10 cents a minute for calls town to town. Currently I use a Cisco ATA 186 to talk to one of my programmers in another town using the FWD gateway. Could I implement the Asterisk software and what recommended hardware to connected these users through the Asterisk head end to place calls and log their activity for charge back? I'm considering having up to 12 VOIP users, possibly 2-3 concurrent users. If I load RH 8.0 what's the minimum hardware requirements -CPU/RAM? Any other suggestions/ ideas would be appreciated. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
