Yep both grandsteam phones are working great! I changed only the ip addresses to match my network. The only soft phone I've tried so far is Kphone. I have it calling both grandstreams - but my workstation (mandrake 10 community) is fighting with my sound card; so I can't talk - listen. I'll try again today to get it going. t o n y On Thu, 2004-05-13 at 03:08, [EMAIL PROTECTED] wrote: > Tony > > Are you able to make this configuration work with 2 sip phone on same Asterisk > server? I am also trying to do the same using xlite softphone abailable on > www.xten.com site. > Please let me know wgat you did? > > Thanks > > Deepak > Quoting Tony <[EMAIL PROTECTED]>: > > > On Sun, 2004-05-09 at 18:51, [EMAIL PROTECTED] wrote: > > > Hi, > > > > > > I've have followed through the help docs in trying to get an initial setup > > > going with two phones and the asterisk server. Firstly, all I'm trying to > > > do is get the two phones actually talking to one another VIA asterisk.. > > > > > > I've added this to sip.conf: > > > > > > [phone1] > > > type=friend > > > host=dynamic > > > defaultip=192.168.1.106 > > > ;username=blah > > > ;secret=blah > > > dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info > > > mailbox=1000 ; Mailbox for message waiting indicator > > > context=sip > > > callerid="Me" <2124> > > > > > > [phone2] > > > type=friend > > > ;secret=blah > > > host=dynamic > > > defaultip=192.168.1.107 > > > dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info > > > mailbox=1000 ; Mailbox for message waiting indicator > > > context=sip > > > callerid="Mini Me" <2123> > > > > > > And in extensions.conf at the very end: > > > > > > [sip] > > > exten => 1,1,Dial(SIP/phone1,20,tr) > > > exten => 2,1,Dial(SIP/phone2,20,tr) > > > exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) > > > > > > These are budgetone 102's, so I've then proceeded to their admin > > interface, > > > and told them that the sip server is: "192.168.1.13". For phone1, all > > I've > > > set is the sip id/username as "phone1" and likewise for "phone2" on phone > > > number two. Rebooted.. But I do not seem to be able to get them to talk > > to > > > asterisk. > > > > > > When issuing a "sip show peers" in asterisk, it displays: > > > > > > Name/username Host Mask Port Status > > > phone2/phone2 192.168.1.107 (D) 255.255.255.255 5060 > > Unmonitored > > > phone1 (Unspecified) (D) 255.255.255.255 0 > > Unmonitored > > > > > > And when a sip show registry is issued, nothing seems to be connected: > > > > > > Host Username Refresh State > > > > > > Could there be something I'm missing in order to get the very basic > > working > > > and then expand on that? > > > > > > Thanks in advance. > > > > > > Matthew > > Matt has made a great start page - > > http://astguiclient.sourceforge.net/scratch_install.html > > Just change the ip's to match your own - you'll be going in minutes! > > t o n y > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ---------------------------------------------------------------- > This message was sent using IMP, the Internet Messaging Program. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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