[Asterisk-Users] Re: Calling one local SIP user from another (using X-Lite)

voice mail. However, if I try to call user2 from user1's X-Lite - or vice-versa - I get a 404 Not Found error. Is there anything obvious that I'm doing wrong? (In particular, do I also need to add entries to extensions.conf for user1 and user2??) Ross. Try adding something similer

[Asterisk-Users] Re: Calling one local SIP user from another (using X-Lite)

this.) Ross Finlayson LIVE.COM http://www.live.com/ ___ 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

[Asterisk-Users] Calling one local SIP user from another (using X-Lite)

I'm having trouble (using a recent CVS Asterisk) calling one local SIP user from another (using X-Lite as the SIP phone). My sip.conf file is as follows: [general] port = 5060 ; Port to bind to bindaddr = IP-address-of-Asterisk-machine ;

[Asterisk-Users] Re: Streaming channels from Asterisk to the Internet

/playSIP/ You can run (e.g.) playSIP -a sip:[EMAIL PROTECTED] | whatever (the -a option means: output the audio stream data to stdout) Ross Finlayson LIVE.COM http://www.live.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http