Asterisk configuration to dail out via SipBroker?
 

Hi Guys,

I can't for the life of me make this work!. :(

It's simple I want an extension in Asterisk to dial out via Sipbroker.

For the test I am trying to make it dial GOOG411.

Extensions.conf

exten => 411,1,Dial(SIP/*18004664411@sipbroker-out/${EXTEN:0}) ;; Dial Goog411

sip.conf

[sipbroker-out]
type=peer
fromuser=xxxxxx@voxalot.com ;; My Account name on SipBroker.
secret=<my password>
fromdomain=shazam.zapto.org ;; My Asterisk Servers Domain name.
host=sipbroker.com
port=5060
canreinvite=yes


I can't for the life of me make this work. I have poured and poured over the SipBroker WiKi on this. But I still can't find the problem.

It's definately the SIP.conf file I believe because I get the following error.

[Jul 28 03:43:43] WARNING[19567]: chan_sip.c:2925 create_addr: No such host: sipbroker-out/411

Any help appreciated!

.-.-.

I have also just tried this simple Extensions.conf entry.

exten => _*.,1,Dial(SIP/sipbroker.com/${EXTEN:0}) ;; * Sipbroker prefix Dialing.

When I try and dial back out with the monkey test. (*266300) I get a little bit of audio and then nothing.

So there must be an issue elsewhere.

I have been able to do this OK with IAX2

exten => 2583,1,Dial(IAX2/cnetguest@projectmf.homelinux.com/17622600) ;; ext for ProjectMF (BlueBoxing)

When I try and use IAX2 with SipBroker I just get a congestion error in the console. :(

.-.-.

-I don't believe SipBroker supports IAX

-Some Asterisk setup suggestions SIP Broker Wiki : Asterisk Configuration

-To make life easy you could add SipBroker as a trunk:

User: anonymous
Pass: <blank>
Proxy: sipbroker.com


The bonus is that any number pushed through SipBroker will undergo an ENUM lookup (all 4 servers I think) as well ... that said I am not sure how you'd set it up to fail the call over to another trunk if there is no ENUM record...

Hi, Yep I'm aware that I can't use IAX to SipBroker.

I was just pointing out that IAX2 out is working but SIP out isn't.

So I am wondering why.

I have opened all the firewall to the Asterisk server so it has me beat.

When placing a call I get about 2-3 seconds of audio and then nothing.

.-.-.

Problem solved. It was one line involving the NAT settings.

localnet=192.168.0.0/255.255.0.0

Now on to the next problem.

Getting PennyTel outgoing working. ;)

.-.-.