I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent to the default demo-thanks. I hang up the call that's waiting. * then calls back 3020 to reconnect the original call. I'm including the progression. astera*CLI> -- Starting simple switch on 'Zap/4-1' -- Executing Wait("Zap/4-1", "1") in new stack -- Executing Answer("Zap/4-1", "") in new stack -- Executing DigitTimeout("Zap/4-1", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("Zap/4-1", "10") in new stack -- Set Response Timeout to 10 -- Executing Dial("Zap/4-1", "Zap/1&SIP/3014&SIP/3016&SIP/3017|35|tr") in new stack -- Called 1 -- Called 3014 Mar 27 17:27:15 NOTICE[10890]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- Called 3017 -- SIP/3017-ba4c is ringing -- Zap/1-1 is ringing -- SIP/3014-556e is ringing -- Zap/1-1 answered Zap/4-1 -- Attempting native bridge of Zap/4-1 and Zap/1-1 -- Attempting native bridge of Zap/4-1 and Zap/1-1 -- Started music on hold, class 'default', on Zap/4-1 -- Playing 'pbx-transfer' (language 'en') -- Stopped music on hold on Zap/4-1 -- Hungup 'Zap/1-1' -- Executing Flash("Zap/4-1", "") in new stack -- Flashed channel Zap/4-1 -- Executing Dial("Zap/4-1", "SIP/3020") in new stack Mar 27 17:27:44 WARNING[10890]: chan_sip.c:1398 create_addr: No such host: 3020 Mar 27 17:27:44 NOTICE[10890]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Timeout on Zap/4-1 == CDR updated on Zap/4-1 -- Executing Goto("Zap/4-1", "#|1") in new stack -- Goto (sip,#,1) -- Executing Playback("Zap/4-1", "demo-thanks") in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup("Zap/4-1", "") in new stack == Spawn extension (sip, #, 2) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Starting simple switch on 'Zap/1-1' Mar 27 17:28:08 NOTICE[10891]: chan_zap.c:5374 ss_thread: Got event 2 (Ring/Answered)... -- Executing Wait("Zap/4-1", "1") in new stack -- Hungup 'Zap/1-1' -- Executing Answer("Zap/4-1", "") in new stack -- Executing DigitTimeout("Zap/4-1", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("Zap/4-1", "10") in new stack -- Set Response Timeout to 10 -- Executing Dial("Zap/4-1", "Zap/1&SIP/3014&SIP/3016&SIP/3017|35|tr") in new stack Mar 27 17:28:09 WARNING[10891]: chan_zap.c:1562 zt_call: Unable to ring phone: Device or resource busy -- Couldn't call 1 -- Hungup 'Zap/1-1' -- Called 3014 Mar 27 17:28:09 NOTICE[10891]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' -- Called 3017 -- SIP/3017-1731 is ringing -- SIP/3014-9afa is ringing == Spawn extension (default, s, 5) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' astera*CLI> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050327/c8710271/attachment.htm