Please help!

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 —>
ACK sip:number-dialed@ SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK-83e61ba2
From: “My Name” sip:1000@;tag=fcaa4b0891c49bb2o0
To: sip:phone-number-dialed@;tag=as1a1d1911
Call-ID: 1a22c640-923799da@
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“15bff7d8”,uri="sip:mynumberagain@",algorithm=MD5,response="ca143a06bd1e4847fe5153a576ff23ce"
Contact: “My Name” sip:1000@
User-Agent: Linksys/SPA942-5.2.2(a)
Content-Length: 0

— (11 headers 0 lines) —
Really destroying SIP dialog ‘1a22c640-923799da@’ Method: ACK

I have 2 NICs in this machine, 1 is assigned an internal 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?


Hi gallagher256,
I have enjoyed Asterisk for 3 year ago.
Perhap, I can help you this problem.
Could you explain more about “But when I go to dial an external number”?
Quoc Viet

When I dial an outside line such as my cell phone number to test with, I get fast busy and the error message above.

Tell us about the hardware u are using. Digium card i mean. Are the drivers properly configured? Since its the problem when trying to call the outside line, that could be a problem with drivers. And you are able to dial and talk between 2 sip phones registered with asterisk rite?

Oh I see…
If that, you need a Voice Card as Digium or Dialogic to connect to PSTN or you must register an account with VOIP Service Provider.
Please look at below link. … +Providers.
Quoc Viet.

I have 6 SIP trunks at
I have them setup in my asterisk machine.
When I try to dialout on SIP I get a fast busy with that debug information above…while dialing from SIP to my cell phone.

I thought that this problem relative to the SIP trunk configuration.
Each VOIP service provider guide us how to set up by their way.
I used to try with so I can not guide you how to solve it.
Please ask some supporters to solve your problem.
Quoc Viet.

Lets’s see your sip.conf and extensions.conf or just post the relevant parts.

I am sure we can get you up and running.