Please give a try like the following, Xlite (Configure Net2Phone A/c) ---> Cisco ASA Firewall ---> Internet cloud,
if the above works then there is no problem with your firewall, replace the nat=yes and canreinvite=yes otherwise you have to allow the ports 5060 (TCP), 5000 to 30000 (UDP) in your router for the Asterisk IP (192.168.0.10) Try this... On Thu, Sep 23, 2010 at 7:33 PM, Alejandro Cabrera Obed <aco1...@gmail.com>wrote: > Dear, I have this scenario: > > - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 > > - Behind a Cisco ASA firewall that connects to Internet > > - SIP trunk to Net2Phone with these parameters (nat=no): > > host=200.58.113.60 > username=DOLLY > secret=123456 > port=5060 > type=peer > dtmfmode=rfc2833 > disallow=all > allow=alaw&ulaw > nat=no > canreinvite=no > qualify=yes > > -Softphones Xlite > > The PBX can't register to Net2Phone, and no calls are made and this is the > log: > > -- Executing [...@macro-dialout-trunk:20] NoOp("SIP/9004-00000008", "Dial > failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") > in new stack > -- Executing [...@macro-dialout-trunk:21] Goto("SIP/9004-00000008", > "s-CHANUNAVAIL,1") in new stack > -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) > -- Executing [s-chanunav...@macro-dialout-trunk:1] > Set("SIP/9004-00000008", "RC=20") in new stack > -- Executing [s-chanunav...@macro-dialout-trunk:2] > Goto("SIP/9004-00000008", "20,1") in new stack > -- Goto (macro-dialout-trunk,20,1) > -- Executing [...@macro-dialout-trunk:1] Goto("SIP/9004-00000008", > "continue,1") in new stack > -- Goto (macro-dialout-trunk,continue,1) > -- Executing [conti...@macro-dialout-trunk:1] > GotoIf("SIP/9004-00000008", "1?noreport") in new stack > -- Goto (macro-dialout-trunk,continue,3) > -- Executing [conti...@macro-dialout-trunk:3] > NoOp("SIP/9004-00000008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: > 20 - failing through to other trunks") in new stack > -- Executing [conti...@macro-dialout-trunk:4] Set("SIP/9004-00000008", > "CALLERID(number)=9004") in new stack > > What can be the problem please ??? > > Thanks a lot > > Alejandro > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thank you with regards, Gopalakrishnan A.N,
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users