For me to it sounds like it could be a firewalling or forwarding problem,
if the Asterisk box is behind a nat router.
I use to forward only these ports:
UDP 5060 (SIP telephony initiation)
UDP 4569 (IAX initiation plus voice)
UDP 10000-10010 (SIP voice carrier)
UDP 10000-20000 is default ports for SIP voice carrier, but I use to reconfigure it to use only 10 ports,
not 10000 as by default.
To receive calls you will also have to have a registration string.
If you go to Asterisk command line and type "sip show registry", you can see if your server has right and
proper registration to be able to receive calls.
Best reg Arne
On 9/5/06, Gary G. Hendershot <[EMAIL PROTECTED]> wrote:
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
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