[asterisk-users] DAHDI Caller ID problem

[Doug Bailey]

----- "Danny Nicholas" <danny at debsinc.com> wrote:

> 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
> 

Assuming that you have standard US CID i.e. a bell fsk spill between the first
and second rings, then you will need to set the following in chan_dahdi.conf :

usecallerid=yes
cidstart=ring
cidsignalling=bell
callerid = asreceived (For incoming trunks)




> [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!


The fact that you get "MWI: Channel 1 message waiting!" indicates that a fsk
spill was processed and contained a message waiting indicator packet.
Unfortunately, it does not indicate that a standard CID packet was included as
well.

>     -- 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'
> 


The error you are seeing ("No start bit found in fsk data") indicates that the
fsk processing code cannot lock onto the fsk spill.  You may want to adjust the
gain applied to the incoming signal while it processes cid.  This can be
adjusted by setting:

cid_rxgain=x.x

This value is in dB and defaults to +5 dB if it is not specified. (You may want
to test both higher and lower values.)


Regards, 
Doug Bailey