Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten => s,1,Verbose(passed id is ${CALLERID(num)}) exten => s,2,Answer exten => s,3,Dial(SIP/100,,) And this is what I'm getting. *CLI> core set verbose 10 Verbosity was 1 and is now 10 -- Starting simple switch on 'DAHDI/1-1' [Sep 17 14:44:05] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 18 (Ring Begin)... [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 2 (Ring/Answered)... [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7706 ss_thread: MWI: Channel 1 message waiting! -- Executing [s at analog:1] Verbose("DAHDI/1-1", "passed id is ") in new stack passed id is -- Executing [s at analog:2] Answer("DAHDI/1-1", "") in new stack -- Executing [s at analog:3] Dial("DAHDI/1-1", "SIP/100,,") in new stack == Using SIP RTP CoS mark 5 -- Called 100 -- SIP/100-b6a22338 is ringing -- SIP/100-b6a22338 answered DAHDI/1-1 == Spawn extension (analog, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' I'm also getting these errors: [Sep 17 14:01:06] ERROR[14462]: callerid.c:562 callerid_feed: No start bit found in fsk data. [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7582 ss_thread: CallerID feed failed: Success [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7686 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' I have tried calleridsignal=dtmf & ring, as well as calleridstart=ring & polarity. No love. I searched on google for info, but nothing I found had a solution for my problem. I know that there's something I missing, but I can't seem to figure it out. Can you all help me? Thanks in advance!
Cidstart=polarity or cidstart=ring will probably fix this. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of herb at cfht.hawaii.edu Sent: Thursday, September 17, 2009 8:04 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] DAHDI Caller ID problem Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten => s,1,Verbose(passed id is ${CALLERID(num)}) exten => s,2,Answer exten => s,3,Dial(SIP/100,,) And this is what I'm getting. *CLI> core set verbose 10 Verbosity was 1 and is now 10 -- Starting simple switch on 'DAHDI/1-1' [Sep 17 14:44:05] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 18 (Ring Begin)... [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7542 ss_thread: Got event 2 (Ring/Answered)... [Sep 17 14:44:07] NOTICE[15308]: chan_dahdi.c:7706 ss_thread: MWI: Channel 1 message waiting! -- Executing [s at analog:1] Verbose("DAHDI/1-1", "passed id is ") in new stack passed id is -- Executing [s at analog:2] Answer("DAHDI/1-1", "") in new stack -- Executing [s at analog:3] Dial("DAHDI/1-1", "SIP/100,,") in new stack == Using SIP RTP CoS mark 5 -- Called 100 -- SIP/100-b6a22338 is ringing -- SIP/100-b6a22338 answered DAHDI/1-1 == Spawn extension (analog, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' I'm also getting these errors: [Sep 17 14:01:06] ERROR[14462]: callerid.c:562 callerid_feed: No start bit found in fsk data. [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7582 ss_thread: CallerID feed failed: Success [Sep 17 14:01:06] WARNING[14462]: chan_dahdi.c:7686 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' I have tried calleridsignal=dtmf & ring, as well as calleridstart=ring & polarity. No love. I searched on google for info, but nothing I found had a solution for my problem. I know that there's something I missing, but I can't seem to figure it out. Can you all help me? Thanks in advance! _______________________________________________ -- Bandwidth and Colocation Provided by api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users