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/[EMAIL PROTECTED] 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/[EMAIL PROTECTED] 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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users