Hello Fellows! I have a TDM2400 and I can't put it to work. Every time it receive a call the Asterisk handle it and call the SIP phone; when people pick up the fone they don't hear nothing and the caller hear the phone rings and nothing happens. In Asterisk console I can see the message answered by the SIP's phone. I lost a lot of time trying to solve this problem without success :(. == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' -- Executing [s at entrada:1] Answer("Zap/21-1", "") in new stack -- Executing [s at entrada:2] Dial("Zap/21-1", "SIP/ramal01&SIP/ramal02&SIP/ramal03|30|tT|r") in new stack -- Called ramal01 -- Called ramal02 -- Called ramal03 -- SIP/ramal03-0070e020 is ringing -- SIP/ramal01-006fd4f0 is ringing -- SIP/ramal02-00705d70 is ringing -- SIP/ramal01-006fd4f0 answered Zap/21-1 == Spawn extension (entrada, s, 2) exited non-zero on 'Zap/21-1' -- Hungup 'Zap/21-1' I got the following message when a enable the usecallerid=yes: Sep 29 16:48:31 WARNING[12369]: chan_zap.c:5961 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/21-1' == Starting post polarity CID detection on channel 21 -- Starting simple switch on 'Zap/21-1' Sep 29 16:48:35 WARNING[12372]: chan_zap.c:5961 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch -- Hungup 'Zap/21-1' I've tested with the Zaptel 1.2/ Asterisk 1.2 and Zaptel 1.4.5.1/Asterisk 1.4.11 and got the same problem. Debian Etch amd64. Thanks for any help! Regards, McCoy Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070929/dbe93b9b/attachment.htm