Hi Here is my sip configuration with fwd. I would recommend getting a fwd account (fwd.pulver.com) as it is free.
; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=reliability register=37526:[EMAIL PROTECTED]/37526 [fwd] type=friend secret=mypassword username=37526 host=fwd.pulver.com My extensions.conf ; at the top you will find the globals section...specify a variable called PHONE1 for sip/fwd..then ; you can change it out or add another phone variable if you have other sip phones. [globals] PHONE1=SIP/fwd ; Extension 1234 ; this will dial my soft sip phone x-lite when someone dials 1234...no ever does though exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." exten => 1234,2,Macro(stdexten,1234,${PHONE1}) exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten => 1236,1,Dial(${PHONE1},30) ; Ring forever exten => 1236,2,Voicemail(u1234) ; Unless busy Also check out John Todd's asterisk conf files. I think they are great and they helped me get my head around all this. http://www.loligo.com/asterisk/current/extensions.conf John Haigh -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly McDonald Sent: Friday, July 04, 2003 8:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Sacrifice? bk, I'm a newbie myself, but I have at least got * working with a sip provider, although the quality was not to my liking, I was hooking up with iconnecthere. Here's what I had in sip.conf: [iconnecthere] type=friend insecure=yes port=5060 username=xyz secret=abc host=natrelay.deltathree.com dtmfmode=inband callerid=15408675512 nat=yes in extensions.conf: exten => 8500,1,Dial(SIP/[EMAIL PROTECTED]) This was just a test so I could dial 8500 and it would call my home phone. Probably have stuff wrong, but it seemed to work. For the rest, extensions.conf has enough stuff in it that you can go and make up your own stuff. HTH, Kelly On Fri, 2003-07-04 at 08:23, BK [address only for mailing lists] wrote: > Hi > > is there any ritual sacrifice a newbie has to perform to be welcome on > this list? > > I am new to this whole PBX thing in general and Asterisk in > particular. > I had hoped that the community on this list would welcome a newbie like > myself and help me with some answers to my stupid questions, but somehow > it seems to me that nobody likes to respond to somebody who appears to > be a complete beginner -- too much bother and a risk to have to explain > everything from scratch -- better not answer at all and all that. > > Well, it may appear that way, but I am not a complete idiot. I know a > lot about mobile switching centres, HLRs, VLRs, IN service nodes, > mediation devices, billing and settlement systems etc -- I just don't > know much about PSTN and PBXes. I would appreciate it if somebody could > help me out with a few hints on how to set up my Asterisk box, in > particular in respect of VoIP as per my last posting. > > thank you very much in advance > kind regards > bk > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Kelly McDonald <[EMAIL PROTECTED]> _______________________________________________ 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