Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that.
So put it in your asterisk knowledge bases. Cheers, have a good weekend ---------- Forwarded Message ---------- Subject: SIP phones can't pick up incoming call on analog trunk - signalling problem? Date: Friday 13 January 2006 13:50 From: C Mylo <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming en default 2 pstn-incoming en default 3 pstn-incoming en default 4 pstn-incoming en default -- Starting simple switch on 'Zap/1-1' Jan 13 13:32:13 WARNING[25905]: chan_zap.c:6105 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Dial("Zap/1-1", "SIP/24|15") in new stack -- Called 24 -- SIP/24-406f is ringing -- SIP/24-406f is ringing -- SIP/24-406f is ringing -- Starting simple switch on 'Zap/4-1' Jan 13 13:32:16 NOTICE[25908]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... -- SIP/24-406f is ringing Jan 13 13:32:18 NOTICE[25908]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/4-1", "SIP/24|15") in new stack -- Called 24 -- SIP/24-c8de is ringing -- SIP/24-c8de is ringing == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- SIP/24-c8de is ringing Jan 13 13:32:21 WARNING[25908]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/24-c8de is ringing Jan 13 13:32:24 WARNING[25908]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/24-c8de is ringing == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' Jan 13 13:32:31 NOTICE[25916]: chan_zap.c:6031 ss_thread: Got event 18 (Ring Begin)... Jan 13 13:32:33 NOTICE[25916]: chan_zap.c:6031 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/4-1", "SIP/24|15") in new stack -- Called 24 -- SIP/24-9f09 is ringing -- SIP/24-9f09 is ringing -- SIP/24-9f09 is ringing -- SIP/23-b60e answered Zap/4-1 Jan 13 13:32:36 WARNING[25916]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 == Spawn extension (pstn-incoming, s, 1) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' & zapata.conf [channels] language=en toneduration=100 context=pstn-incoming signalling=fxs_ks usecallerid=yes hidecallerid=no usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no flash=100 transfertobusy=yes busydetect=yes busycount=4 busypattern=400,400 callprogress=yes progzone=au ringtimeout=6000 faxdetect=incoming musiconhold=default channel=>1 ;X100P channel=>2 ;X100P channel=>3 ;X100P channel=>4 ;X100P & zaptel.conf fxsks=1-4 loadzone = au defaultzone=au & extensions.conf ...relevant part... [pstn-incoming] exten => s,1,Dial(SIP/24|15) exten => s,n,Dial(SIP/21&SIP/22&SIP/23&SIP/24&SIP/25&SIP/26&SIP/27|120) My experience with asterisk isn't with analog signalling and it has been a long time since I looked at zapata.conf with a keen eye. I've never not been able to pick up a call so I'm a little baffled and that's why i'm presuming it's something to do with signalling. Thanks ------------------------------------------------------- _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users