Hi all. I am unable to answer calls coming into asterisk over PSTN. (UK) I want to have a call answered by my TDM400P/FXO module and forwarded to a sip phone. When I make a call from the PSTN to the BT line installed on my FXO module the sip phone rings however, when i pick up the call using the sip phone, the incoming call is not answered/routed by asterisk. As a result the sip phone is left hanging and the incoming call remains unanswered. my zapata.conf now looks like this. ------------------------------------------------- ; Configuration file ; [channels] language=uk group=1 context=from-pstn usecallerid=no cidstart=polarity signalling=fxs_ks channel => 4 ------------------------------------------------- debug info ------------------------------------------------- *CLI> == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' -- Executing NoOp("Zap/4-1", "--- calling on 01189xxxxxxx (s) ---") in new stack -- Executing Dial("Zap/4-1", "SIP/1001|20") in new stack -- Called 1001 -- SIP/1001-5c18 is ringing -- SIP/1001-5c18 answered Zap/4-1 May 24 11:12:35 WARNING[32757]: chan_zap.c:3646 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 == Spawn extension (from-pstn, s, 2) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' --------------------------------------------------- All the other variations of my configuration works well, it is just this part. Any help much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050524/e3a4dbae/attachment.htm