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you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing<br>what are the contexts you are using with your peers?<br>what is the dial plan triggered when calling your destination number?<br>--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
<br><br><br><br>> Date: Sun, 25 Oct 2009 15:19:28 +0100<br>> From: robert.bielik@xponaut.se<br>> To: asterisk-users@lists.digium.com<br>> Subject: [asterisk-users] SIP interconnection problem<br>> <br>> Hi all,<br>> <br>> I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using<br>> IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a <br>> Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension<br>> on the other * I get a "Failed to authenticate on INVITE" on the * to which the Zoiper is registered:<br>> <br>> -- Accepting AUTHENTICATED call from 192.168.10.113: << Zoiper IP<br>> > requested format = gsm,<br>> > requested prefs = (),<br>> > actual format = ulaw,<br>> > host prefs = (ulaw|alaw|gsm),<br>> > priority = mine<br>> -- Executing [010001@users:1] Dial("IAX2/2200-12940", "SIP/010001@192.168.10.11") in new stack<br>> == Using SIP RTP CoS mark 5<br>> -- Called 010001@192.168.10.11 << Other *<br>> [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200@192.168.10.77>;tag=as3e4fedb8' << 192.168.10.77 == * for Zoiper<br>> -- SIP/192.168.10.11-0a1716f8 is circuit-busy<br>> == Everyone is busy/congested at this time (1:0/1/0)<br>> -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION'<br>> -- Hungup 'IAX2/2200-12940' <br>> <br>> Why does * try to authenticate on sip:2200@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on<br>> the IAX phone (not sure this has any meaning in IAX at all)<br>> <br>> Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ?<br>> <br>> TIA<br>> /R<br>> <br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> <br /><hr />Windows 7: I wanted more reliable, now it's more reliable. <a href='http://microsoft.com/windows/windows-7/default.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009' target='_new'>Wow!</a></body>
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