Yes you should be able to make/get calls using a SIP phone against an IAX2 provider connection ... I have done IAX2 against Teliax and it worked fine, but I ended up switching to Sip because the sound quality seemed better.
Your problem sounds like it might be a firewall issue. Am guessing but think I might be correct. Questions to consider: 1) Is your Aslinux box in PBX only mode (not used as a firewall/router)??? If so, the problem may be at your firewall. You may be filtering the IAX2 protocol. Suggest you use the setup on your firewall (and any physical connections required) to make your Astlinux box the "DMZ host". In most SOHO (cheap) firewall/routers, this will have the effect of opening every port in the universe up to your Astlinux box. NOT A GOOD THING. But will allow you to test your setup without the firewall interfering with the session. If it works like this, then know it was indeed a firewall issue, so you know what to fix. The fix involvolves putting the Astlinux box back behind the firewall and setting up "port forwarding" on your firewall/router to get the right port directed to right private IP address. I have seen some SOHO firewall/routers that did a pretty poor job of port forwarding. In some cases, the only way I could get Asterisk to work was to put the pbx in the dmz. This is risky and is not recommended but it makes the implementation simple and normally works. Better to either use Astlinux as your firewall/router or buy a firewall/router that will do a proper job of port forwarding (m0n0wall, pfsense, IPCop, others). 2) if your Astlinux box is also your firewall/router, the required ports for IAX2 should be open to the outside world by default (assuming you set it up properly). Would expect you might have trouble with default config only if you were talking to provider using SIP. However, you should check and make sure you have listed the following ports as being open to support both IAX2 and SIP to the outside world ... Tcp 5060 Udp 2727, 5060, 4569, 5036, 9999 >< 20001 3) it may be that your problem with incoming calls is that Teliax cant find you. This is common with SOHO installations where there is a dynamic IP address being used. In these cases, it is best to get an account with a provider of dynamic dns service. There are a few choices and they all work. DYNDNS.ORG is one of the most popular. Once you have this handled, you will have to tell your firewall/router to register with the dynamic dns outfit and setup your iax.conf to tell Teliax where you are. Good stuff about this on the voip-info.org wiki. G.Hendershot -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Knospler Sent: Tuesday, September 05, 2006 1:26 PM To: [EMAIL PROTECTED] Subject: [Astlinux-users] IAX2 to SIP I have a Teliax IAX2 configured line and a Polycom SIP phone working in Asterisk. I can make outgoing calls fine, but incoming calls direct to the sip phone result in a protocol conversion error.. The phone rings, but the call is dropped when answered. Asterisk console lists protocol Requests type 256 and phone is type 4 or 8. Should I be able to make calls in both directions when using a sip phone and a iax2 line? -Rick _______________________________________________ Astlinux-users mailing list [email protected] http://lists.kriscompanies.com/mailman/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED] _______________________________________________ Astlinux-users mailing list [email protected] http://lists.kriscompanies.com/mailman/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED]
