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 [EMAIL PROTECTED]:1] Answer("Zap/21-1", "") in new stack
-- Executing [EMAIL PROTECTED]: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
_______________________________________________
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--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