On Mon, Oct 20, 2008 at 12:23 AM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > The second call its OK, so the problem it is just with the Dial(SIP/102), so > try: > originate SIP/102 application Dial SIP/102 > and > originate SIP/101 application Dial SIP/102 > and > originate SIP/102 application Dial SIP/101
ns1*CLI> originate SIP/102 application Dial SIP/102 ns1*CLI> == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/102-0824a330 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-0824a330 requested special control 16, passing it to SIP/102-082256c0 -- Started music on hold, class 'default', on SIP/102-082256c0 -- SIP/102-082256c0 answered SIP/102-0824a330 -- Packet2Packet bridging SIP/102-0824a330 and SIP/102-082256c0 -- Stopped music on hold on SIP/102-082256c0 ns1*CLI> originate SIP/101 application Dial SIP/102 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/102) on SIP/101-08249e28 == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-082256c0 is ringing -- SIP/102-082256c0 answered SIP/101-08249e28 -- Packet2Packet bridging SIP/101-08249e28 and SIP/102-082256c0 ns1*CLI> originate SIP/102 application Dial SIP/101 == Using SIP RTP CoS mark 5 -- Launching Dial(SIP/101) on SIP/102-08254038 == Using SIP RTP CoS mark 5 -- Called 101 -- SIP/101-08252a40 is ringing -- SIP/101-08252a40 answered SIP/102-08254038 -- Packet2Packet bridging SIP/102-08254038 and SIP/101-08252a40 So I the two extensions are able to call each other with the later two sets of commands so there is hope :-). Would my NAT have anything to do with it since I'm specifying the proxy host that is outside of my firewall? _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users