[Asterisk-Users] Can PSTN CallerID be fowarded to a SIP phone extension?

Walt Reed asterisk at linuxguy.com
Fri Aug 20 07:24:33 MST 2004


On Fri, Aug 20, 2004 at 08:48:54AM -0400, James Freire said:
> Sorry about this. I forgot to include the error from the CLI upon recieving an incomming call.
> 
> Aug 20 08:37:53 NOTICE[622610]: chan_zap.c:5053 ss_thread: Got event 2 
> (Ring/Answered)...
>     -- Detected ring pattern: 338,0,0
> Aug 20 08:38:00 WARNING[622610]: chan_zap.c:5124 ss_thread: CallerID 
> returned with error on channel 'Zap/8-1'

OK, this gets to the root of the problem. It may be that * is having
difficulty decoding the callerID info. CalledID is sent via FSK encoding
- it's like a little blip of modem noise between the first and second
ring.

Some people have reported success playing with the
rxgain zapata setting...

I can't really help any further as it sounds like it's probably a line
problem (noise, weak levels, impedance problem, etc.)

Do you have muliple lines that this is occuring on or just one? How is
your wiring from * to your NID? Old? Split off a dozen times?

Note that commercial callerID units can sometimes be better at callerID
decoding - they are doing this in hardware where * does it in software.




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