Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: riccge on November 06, 2006, 07:04:40 PM
-
My SME is behind a router with port 5060 and range 10000-20000 forwarded,so outgoing calls with an italian carrier (Skypho) and with Freecall work without problems.
The blackhole is where are the incoming calls from geografic number that i have on Skypho?
After adding into the trunk "nat=yes" if i call my geo_number it sounds busy.
Probably need i to setup 2 trunks:inbound and oubound?
Any suggestions?
Thanks
-
After adding into the trunk "nat=yes" if i call my geo_number it sounds busy.
Probably need i to setup 2 trunks:inbound and oubound?
OK, you shouldn't need nat=yes, it does something that your carrier should be doing anyway. Neither do you need two trunks (at least, you shouldn't). SAIL has no concept of an inbound trunk (at least for SIP operations - N.B. IAX2 is different).
Asterisk needs no special sip.conf code to answer an incoming SIP call. As long as there is a correct entry in extensions.conf in the inbound context then the call will be accepted. So there are two posiible conclusions...
Either your carrier is delivering to the wrong IP address or your asterisk, or firewall is rejecting the call.
Assuming your ports are correctly forwarded and the carrier has the correct delivery address then it is highly likely that you have the wrong entry in extensions.conf for your carrier and so the call is being rejected as "404 notfound" by asterisk. When this happens, you see nothing at the asterisk console unless you are running sip debug.
Now, some carriers deliver calls against the geographic number they give you, or they may deliver against the E164 version of it. Others deliver the call against the account number they give you. There is no standard but you can usually work out which it is from the carrier's asterisk set-up documentation. If not you will have to do a bit of detective work...
Turn on sip debugging at the asterisk console with...
sip debug
Now run your call. This will show each SIP packet arriving and what is in it, - try not to do this in a busy SIP environment or you will just get too much data to digest!
This should put you onto the right track. Once you know why your calls are being rejected it's usually a simple step to correct the error.
Turn sip debugging off with...
sip no debug.
Kind Regards
Selintra
-
What error do i have to find?
Something particular?
I think it's not a good idea to post part of debug...
-
What error do i have to find?
Something particular?
I think it's not a good idea to post part of debug...
I found only a Destroy call but still don't understand why?
-
What error do i have to find?
Something particular?
I think it's not a good idea to post part of debug...
I found only a Destroy call but still don't understand why?
The Mistery is solved!!
insecure=very
this was the answer!
...but only trying becuse in debug mode I couldn't see any error...
-
You solved your problem - Cool!
There should have been an "unauthorised" reply to the SIP invite/challenge in your sip debug. I am a little surprised you didn't see it.
There is a write-up here on creating carriers...
http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter259
It covers insecure=very and the other tuples you may need in your SIP peer.
Kind Regards
Selintra