[asterisk-users] CallerID Error causes problems for Polycom phones

[Lee Jenkins]

Hi everyone,

I have been dealing with a certain issue with a particular customer site
for months now.  The problem occurs when there is an error with caller
id as shown in the following:

WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error
on channel 'Zap/3-1'

When this happens, it appears that the call still goes through as I can
see the caller still navigating through the systems menus and dialplan 
by watching the CLI.

The problem however is manifested with polycom 301's that are setup with
the system.  When a call comes in after receiving that particular caller
id error, the polycoms, which are on a group ring by the way, will all
ring but you cannot pickup the call.  The Answer|Reject soft buttons
display, but only the reject button works.  Pressing the Answer button
or picking up the handset does nothing.

Since only the "Reject" button works someone has to go to each phone and
hit the reject button (4 polycoms in this department) so the ringing
will at least stop.

It's been about 3 months tracking this problem down (even drove the 2.5
hours back and forth to replace the sangoma card to try to fix the
problem) and the customer is about ready to have me pull the system
because of it.

I can easily reproduce the problem with Polycom phones (but not the
actual error). Just issue a .call file using the local channel calling
one number and having the call bridged to a polycom phone (at least
301's here):

Action: Originate
Channel: local/111 at Management
Context: to_meetme
Exten: s
Priority: 1
Async: true

The above will cause the polycom to exhibit the behavior mentioned 
above.  However, sending a .call file like the following causes the 
phone to work as it should:

Action: Originate
Channel: local/111 at Management
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=1234|CALLERID(name)=Homey D Clown
Async: true

I also have tried this with Aastra, Grandstream and XLite soft phones 
and they do not exhibit the same behavior.  Instead these other phones 
simply show the default caller id info as set in sip.conf and allow you 
to answer them.

Any help or suggestions would be greatly appreciated.

OS: CentOS 4
Asterisk: 1.2.17
Sangoma A200 with 2 fxo ports.

-- 
Warm Regards,

Lee