All,
Got it
working. Turned out to the cable between the out port on the tdm400 and the
telephone wall socket. It appears that it requires a cable that you would
ordinarily get with a modem. e.g. two wires (red & green) with the red wire
on the right if you look at the rj11 with the lever at the top (or the red cable
on the left if look from the bottom of the rj 11 plug , with the copper pins
exposed)
Hope
this helps someone.
D.
Confidentiality Notice: The information contained in this e-mail is for the intended recipient(s) alone. It may contain privileged and confidential information that is exempt from disclosure under English law and if you are not an intended recipient, you must not copy, distribute or take any action in reliance on it. If you have received this e-mail in error, please notify us immediately either by using the reply facility on your e-mail system or by contacting us at [EMAIL PROTECTED] . If this message is being transmitted over the Internet, be aware that it may be intercepted by third parties.-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: 15 June 2005 20:08
To: [email protected]
Subject: [Asterisk-Users] phantom answerPeople,My goal is to get asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the phone, The TDM400p is installed and working. (See below) When ever I dial a number that is directed to the outgoing port on my card (fxs/fxo?) I get no ringing, then it claims its been answered. the CLI reports the following:Executing Dial("SIP/301-f97a", "Zap/4/01614299100|20") in new stack
-- Called 4/01614299100
-- Zap/4-1 answered SIP/301-f97a
Jun 15 17:57:38 NOTICE[11121]: rtp.c:277 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.7
-- Hungup 'Zap/4-1'Anyone Any Ideas? BTW Apologies for the disclaimer at the bottom, but the mail server adds it on by default and there's nothing I can do about it.*CLI> zap show channels
Chan Extension Context Language MusicOnHold
pseudo default
1 default default
4 incoming default
*CLI>This is the important bit from zapata.conf
; DYLAN ADDED FROM DIGIUM.COM ********************************
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=yes ; Asterisk trains to the beginning of the call, number is in milliseconds
callerid=01614830073
signalling=fxo_ks
group=1
context=default ; Points to the default context of your extensions.conf
channel => 1signalling=fxs_ks
;callerid=asreceived
group=2
context=incoming
channel=> 4
; END OF DYLAN ADDED FROM DIGIUM.COM *************************Confidentiality Notice: The information contained in this e-mail is for the intended recipient(s) alone. It may contain privileged and confidential information that is exempt from disclosure under English law and if you are not an intended recipient, you must not copy, distribute or take any action in reliance on it. If you have received this e-mail in error, please notify us immediately either by using the reply facility on your e-mail system or by contacting us at [EMAIL PROTECTED] . If this message is being transmitted over the Internet, be aware that it may be intercepted by third parties.___________________________________________________________________ This message has been scanned by the Datanet MessageScreen Service. For more information please visit http://www.MessageScreen.co.uk
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