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 On Sun, Oct 19, 2008 at 11:46 PM, Stephen Reese <[EMAIL PROTECTED]> wrote: > On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez <[EMAIL PROTECTED]> > wrote: > > Stephen: > > Your configuration files looks fine. Try from the CLI issuing "originate > > SIP/101 extension [EMAIL PROTECTED]", having the 101 online, then do that > > with > > "originate SIP/102 extension [EMAIL PROTECTED]". See what happens. > > If you got a CVS commit, commit again or try installing a release. > > http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz(for > > download) > > Regards, > > Juan > > I grabbed the latest tarball and installed it. > > The extension rings through to 101 and then when I answer it tries to > ring through to 102 but seems to fail. > > ns1*CLI> originate SIP/101 extension [EMAIL PROTECTED] > == Using SIP RTP CoS mark 5 > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/101-08245390", > "'SIP/102',20") in new stack > [Oct 19 23:41:40] WARNING[20305]: channel.c:3470 ast_request: No > channel type registered for ''SIP' > [Oct 19 23:41:40] WARNING[20305]: app_dial.c:1450 dial_exec_full: > Unable to create channel of type ''SIP' (cause 66 - Channel not > implemented) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [EMAIL PROTECTED]:2] Hangup("SIP/101-08245390", "") in new > stack > == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08245390' > > The extension rings through to 102 and when I answer the line it > begins to ring line 101. > > ns1*CLI> originate SIP/102 extension [EMAIL PROTECTED] > == Using SIP RTP CoS mark 5 > -- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-08249e28", > "SIP/101&SIP/[EMAIL PROTECTED],30") in new stack > == Using SIP RTP CoS mark 5 > -- Called 101 > == Using SIP RTP CoS mark 5 > -- Called [EMAIL PROTECTED] > -- SIP/101-08244e88 is ringing > -- SIP/vitel-outbound-0825d1e0 is making progress passing it to > SIP/102-08249e28 > -- SIP/vitel-outbound-0825d1e0 is ringing > -- SIP/vitel-outbound-0825d1e0 answered SIP/102-08249e28 > -- Packet2Packet bridging SIP/102-08249e28 and > SIP/vitel-outbound-0825d1e0 > == Spawn extension (default, 101, 1) exited non-zero on 'SIP/102-08249e28' > > I'm at a loss. Thanks for your help. > -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227
_______________________________________________ -- 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
