I assume you get a dial tone when you pick up the handset? If you had a good phone-to-asterisk connection, debug would show a connection or rejection when you did 99#.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones added that line to the extensions.conf file because i could find a way to add it in the GUI. I put it under the dial plan that i have selected. i just get a busy signal i tried #99 just 99, *99 nothing works. debugging isnt showing anything. _____ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 13 Jul 2009 12:12:16 -0500 Subject: Re: [asterisk-users] setting up phones Most folks (AFAIK) use TFTP to connect to the Asterisk server. I personally use HTTP, but that took a few days of research to figure out. You're really only using that protocol for configuration and log transfers. The actual lifting is done on a TCP or UDP connection. Your posts Friday indicated that Asterisk was up and "functional" but that you couldn't make your phones talk to it. I'm thinking that instead of trying to dial phone-to-phone, that you should first make one phone talk to asterisk using this little snippet. - exten => 99,1,Playback(tt-monkeys) - exten => 99,2,Playback(vm-goodbye) - exten => 99,3,hangup When you get your phone where it can dial 99 and get a message, you will be ready to proceed with P2P talking. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose Sent: Monday, July 13, 2009 12:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] setting up phones Ok here is what i did. reinstalled asterisk (i used the make samples option) and asterisk-gui in the gui i did the following created a dial plans using the defaults. no outgoing dial plans just local crated two users logged into the web interface with each phone and pointed them to our asterisk server. Just the Proxy server and Registrar server. Still doesn't work. Should i be able to use the configuration server settings form the phones web gui. it has the options for tftp, ftp, http, https. I don't know how this is supposed to be configured. I still don't know what the problem is and sip set debug off does display any info like it was lastweek. I am just trying to use the gui like you suggestd > Date: Fri, 10 Jul 2009 14:22:25 -0700 > From: asterisk....@sedwards.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] setting up phones > > On Fri, 10 Jul 2009, Ott Rose wrote: > > > I don't think the GUI is editing the conf files correctly. I am not sure > > I have configure things right. At this point i think i am going to start > > from scratch. > > Yea! > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _____ Windows LiveT HotmailR: Spread the word when you add celeb photos to your e-mails. Check <http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_ WL_QA_HM_celebrity_photos1_072009&cat=celebrity> it out. _____ BingT brings you health information from trusted sources. Try it now. <http://www.bing.com/search?q=pet+allergy&form=MHEINA&publ=WLHMTAG&crea=TXT_ MHEINA_Health_Health_PetAllergy_1x1>
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