I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls recently. 2 weeks ago, I upgraded to CVS HEAD: Asterisk CVS-HEAD-02/21/05-09:07:50 Still didn't make or receive calls to FWD since the upgrade, but everything else has worked flawlessly (including sixTel, NuFone, etc.). All my softphones (SIP and IAX2) and Sipura-2000's work perfectly too. On to the problem... A few days ago, I signed up for an account with SIPPhone. When I did a "sip reload", which had the register statement, I immediately got a call "welcoming" me, so I thought everything was fine. It wasn't. I have been unable to make any calls to sipphone, and even though the registration appears to work (and my.sipphone.com shows me as "online"), all calls to my number actually claim that I am unavailable, and go directly to voicemail. Before I show my configs and CLI output, a few more background data points: I can successfully connect to sipphone with their own download of X-Lite (pre-configured), and I can set a profile in SJPhone by hand and it works too, both incoming and outgoing, so I have the correct password, etc. Today, I tested outgoing calls on FWD (actually to use the peering to test incoming on sipphone), and my calls to FWD are failing now as well. Incoming from FWD (via IAX2) still works correctly. Worse, I also tried to go back to SIP-based outgoing to FWD, and I get the same error as I do for sipphone, so now I am starting to suspect that it's Asterisk CVS HEAD that's possibly the problem... Finally, the machine that is connected to both FWD and SIPPHONE is on a public static IP address, so there are no NAT issues involved here, and no STUN services needed either. OK, here is the sip.conf entry: register=1747XXXXXXX:[EMAIL PROTECTED]/4321 [proxy01.sipphone.com] type=peer ;auth=md5 secret=YYYYYYY username=1747XXXXXXX fromuser=1747XXXXXXX fromdomain=proxy01.sipphone.com host=proxy01.sipphone.com nat=no qualify=no canreinvite=no disallow=all allow=ulaw ;context=default ;callerid="Hadar Pedhazur" <1747XXXXXXX> (The above has been variously named sipphone, sipphone-out and now proxy01.sipphone.com, all with the same exact result! Also, the above has been tried with auth=md5 uncommented as well, and also no password, and insecure=vary, etc.) Now extensions.conf: ; Dial SIPPhone with a prefix of 76 exten => _76.,1,SetCallerID(${SIPPHONENUM}) exten => _76.,2,SetCIDName("Hadar Pedhazur") exten => _76.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) OK, here's the output from a call: -- Called [EMAIL PROTECTED] -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)" back from 198.65.166.131 -- SIP/proxy01.sipphone.com-78d5 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Notice that at the end of the "Got SIP response" line, is the correct IP address of their server, so it's finding the correct server. As mentioned above, if I switch FWD to call via SIP, I get the same _exact_ error message, but from FWD's correct IP address rather than SIPPhone. This seems very suspicious to me... Finally, just for completeness, here is the CLI output for attempting to call FWD via IAX2. This used to work, though I can't say when it started failing: -- Called fwd-gw/612 -- Call accepted by 65.39.205.121 (format ulaw) -- Format for call is ulaw -- IAX2/fwd-gw-4 is busy I have called _many_ times, and every time I get an instant "is busy" in the CLI, and I can receive calls without a problem, so I don't think it's that they really are busy. For now, I'm more interested in fixing the SIPPhone problem, and if that ends up working, and doesn't shed light on the FWD problem, I'll move on to that. Of course, PITA that it would be, my next move if no one here can help will be to restore my settings from a few weeks back (yes, I back up religiously :-), and see if 1.0.3 will "just work". Thanks in advance to any kind soul who has some insight! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users