Hi guys, I’m very new but I think I’ve atleast got the hang of SOME Of this wonderful PBX software called Asterisk…but alas, I’m having HELL with it, probably due to by noobness.
For the life of me I simply CAN NOT get SIP calls (inbound or outbound) to go through, no matter what.
I have a very high end machine running CentOS 5 and Asterisk 1.4.15, I am able to register extensions on the PBX and call between them, as well as do the echo test and various other “default” things that come with Asterisk. But when I go to dial an external number, I just get a fast busy and this shows in the debug:
<— SIP read from 192.168.1.107:5060 —>
ACK sip:firstname.lastname@example.org SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK-83e61ba2
From: “My Name” sip:email@example.com;tag=fcaa4b0891c49bb2o0
CSeq: 102 ACK
Proxy-Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“15bff7d8”,uri="sip:firstname.lastname@example.org",algorithm=MD5,response="ca143a06bd1e4847fe5153a576ff23ce"
Contact: “My Name” sip:email@example.com:5060
— (11 headers 0 lines) —
Really destroying SIP dialog ‘firstname.lastname@example.org’ Method: ACK
I have 2 NICs in this machine, 1 is assigned an internal 192.168.1.102 address, the other is assigned my real public routable IP address, I have 5 total usables, the internal IP also goes through a router which has a public ip as well.
I would like all extensions to register via the private network and transfer all the sip calls to the public network via the 2nd NIC in the asterisk machine…thus eliminating the need for NAT on the SIP dialing side of asterisk. I’m kind of assuming my dialplan is screwed up but I’m not totally sure.
Does anyone have any ideas and/or some sample configs that would allow this setup?