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Dear all, I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have trouble, I see on XLITE console: Registration Error: 503 - Service unavailable. Someone have a tutorial or a step by step description how to do that ? Thanks in advance -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/83d6d4ff/attachment.htm

Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: *Registration error: 404 Not found* Here my configuration file of asterisk: *[root at dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[root at

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten