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. ;) .-.-. |
All times are GMT. The time now is 10:13 AM. |
Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.