We recently updated one of our sites to the newest 1.4 Asterisk release (1.4.29 as of the time of the upgrade).  So far things have been working fairly well.  We had a convoluted queues.conf setup in the past with all of the queue members (incorrectly) entered in that file.  Calls would ring to the correct phones, but there was no way for an unused phone to be logged out.  It also made management cumbersome.  With the new upgrade, calls would still route the same but users could not answer the call.  If they picked up the call they ended up with a “dead air” connection to nothing while on the caller side it continued acting like nobody picked up, and eventually the call would timeout and go to voicemail.  Very strange.  We switched to having an easier to manage dynamic setup where phones can log in and out to different call queues and I added some fancy (I think) features that I’ll outline in my next post, which handles our specific call routing needs.

Aside from that, we seem to be having mystery problems of voicemails suddenly not always saving.  Debugging is now activated and I hope I can track down the problem quickly.