Hi Ott, Have you made it work with Asterisk and Aastra IP Phone. I am also trying the same thing, in Asterisk it shows registered OK but when I dial from extension to extension, call is failed...
Please let me know have you made it work...:( On Mon, Jul 13, 2009 at 11:46 PM, Ott Rose <sixfourimp...@hotmail.com>wrote: > > I did " set sip debug on " from the CLI > > It doesn't scroll messages like it did on Fri > > > i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which > isn't either one of the ips of the asterisk server. then it hung up > > i do have a dial tone > > > i just figured something out after reading my post. > > > if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to > the extension and the other phone rings. > > still can't get the 99 to call the asterisk server to work i put in the ips > of the server but it hangs up right away > > ------------------------------ > From: da...@debsinc.com > To: asterisk-users@lists.digium.com > Date: Mon, 13 Jul 2009 12:57:59 -0500 > > Subject: Re: [asterisk-users] setting up phones > > 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 Live™ Hotmail®: Spread the word when you add celeb photos to your > e-mails. Check it > out.<http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_WL_QA_HM_celebrity_photos1_072009&cat=celebrity> > > > ------------------------------ > > Bing™ 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> > > ------------------------------ > Bing™ 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> > > _______________________________________________ > -- 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 > -- Thank you with regards, Gopalakrishnan A.N,
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