Re: [asterisk-users] Rookie / sip and extensions
7 jul 2012 kl. 21:07 skrev Mikhail Lischuk: Thomas Perron писал 07.07.2012 21:48: exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 I dont know what you are trying to do, but: 1) Peer doesn't have to be the same name as context. Change [incoming] in sip.conf to something like [voipvip] - it will be easier later when you have more peers. 2) What is 16175551212 ? You don't have such peer in sip.conf. If it's a number, Dial should be SIP/peer/number, for example SIP/voipvip/617 or whatever you want to dial 3) If you've posted your real password here - I strongly suggest you change it right now Please note that the account name is the name within square brackets. The username= option (now renamed to defaultuser= ) is a very different thing, and NOT the username of the account. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rookie / sip and extensions
Sorry for blasting another desperate note but I am trying! I have changed the username and password and IP to protect my system. But, the logic is unchanged. It is does not work! I simply want to dial the telephone number provided to me for my DID which corresponds with my SIP info. And, then it should connect and hit the incoming context and simply dial the 617 number. I am close but no cigar. Now I get a fast busy tone only. What is missing or what is needed please? extensions.conf [globals] ; ; [incoming] ; ;exten= s,1,Goto(125010155_incoming) ; ;[125010155_incoming] exten = s,1,Answer exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 type=peer secret=funnytiger nat=auto insecure=invite,port host=69.90.209.11 fromdomain=69.90.209.11 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc srvlookup=yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rookie / sip and extensions
On Sat, Jul 7, 2012 at 1:48 PM, Thomas Perron thomas.per...@gmail.comwrote: extensions.conf [globals] ; ; [incoming] ; ;exten= s,1,Goto(125010155_incoming) ; ;[125010155_incoming] exten = s,1,Answer exten = s,n,Dial(SIP/16175551212) sip.conf [general] ;register = 125010155:funnyti...@sip3.voipvoip.com/125010155 register = 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 type=peer secret=funnytiger nat=auto insecure=invite,port host=69.90.209.11 fromdomain=69.90.209.11 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc srvlookup=yes If these are actual copy / pastes from your extensions.conf and sip.conf files, with just passwords changed, your issue probably comes from your over abundant use of semi-colons (;) at the start of several lines. The semi-colon indicates a comment line to the asterisk parser, and thus isn't parsed. Your only exten = line in your [incoming] context is commented out, as is the name of your [125010155_incoming] context, and your first register statement. Set the CLI verbosity to 6 (core set verbose 6) and then try to dial in again, and paste the failed output as a response to this email, and we can diagnose from there. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
