Hello, patching v1.0.5 on my system removed the problem for me. But yes it seems strange that this feature was inserted into a final release with very little documentation of the wide implications that are caused by the change.
This was corrected in CVS with the addition of a diabling flag for the dial command, but maybe this is a message that we should start an official beta release period before a release so that people can test pre-releases even for just a week to report problems before it is unleashed upon the world as an official release MATT--- -----Original Message----- From: Mark Eissler [mailto:[EMAIL PROTECTED] Sent: Friday, February 04, 2005 9:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Callerid problems with 1.0.5 Yikes! On Feb 4, 2005, at 1:26 PM, Jay Milk wrote: > Can someone clarify what's going on here? > > I'm running 1.0.5, and I see caller-id come through just fine from one > extension to the other, as well as for incoming and outgoing calls > (iax2). What are you folks seeing there? > The behavior that was reported by Kevin is/was exactly the same behavior that I was experiencing with 1.0.5 and reported in another thread. I switched back to 1.0.2 to resolve that problem and another I was experiencing (SIP calls ringing forever instead of disconnecting even when voicemail had already picked up). Reading through the bug tracker on this one I must say I'm a bit confused. I understand the concept of showing useful/relevant callerid when a call is transferred (from park or some other extension) but I don't understand why a call should ever show the recipient extension's callerid. My understanding is that this is the default behavior when no other callerid is present and for some reason inbound callerid is getting wiped out because it's not correct. That some people are experiencing problems with this while others are not leads me to believe that it might be a problem that is exacerbated depending upon the dialplan setup. I'm just thinking this at the top of my head now, haven't looked back at my dialplan yet. What's annoying, either way, is that when this change was made the behavior of existing, functioning setups broke. I don't recall seeing any documentation for 1.0.5 that noted this might be the case and if the documentation is lacking...well, that's a problem. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
