James Mutuku
2009-Sep-22 14:12 UTC
[asterisk-users] setting up a IP based voip carrier account
Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc is circuit-busy Are there any settings I am leaving out? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090922/031ef104/attachment.htm
Alex Balashov
2009-Sep-22 15:04 UTC
[asterisk-users] setting up a IP based voip carrier account
James Mutuku wrote:> Hellos, > > My voip carrier has assigned me a IP based account...where they only > give me the IP to call through. I have setup the dial plan > > exten => _7XXX.,1,Answer() > exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) > exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) > exten => _7XXX.,4,Hungup() > > > Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error > > SIP/Y.Y.Y.Y-35dc is circuit-busy > > Are there any settings I am leaving out?No; the "circuit-busy" error is due to negative SIP feedback that peer is sending you; perhaps a 404 Not Found or a 503 Service Unavailable or something of the sort. To find out what the essence of the problem is, you may need to turn up verbosity (core set verbose 60) on the CLI. If that doesn't divine it, do a packet capture: tcpdump -i ethX -A -s 0 -n "udp port 5060 and host Y.Y.Y.Y" Some consider it good practice to add a SIP peer in sip.conf for that endpoint regardless. You don't have to provide any authentication information (username, secret) in the peer definition. This will also be necessary - not optional - if you plan to accept incoming calls from the same carrier, so that you can set it not to challenge the carrier for authentication credentials: insecure=port,invite and to route the calls into a particular dial plan context: context=incoming-calls -- Alex -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671