Trying out 10.0.0-rc1. It dies starting up:
== Parsing '/etc/asterisk/codecs.conf': == Found
[Nov 11 17:07:05] WARNING: translate.c:1060
__ast_register_translator: empty buf size, you need to supply one
[root@asterisk ~]# Where do I supply the "buf size" to the translator? And what should it be?? sean
Version 2.1 of app-espeak and app-flite modules just got released.
Changes include voice selection for flite, support for 16kHz file
playback in asterisk 1.6.x, use of libsamplerate for faster resampling,
better error handling and cleaner code. You can get them here: Asterisk espeak module:
http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module:
If you are providing a hosted phone system to customers, how do you deliver the calls?
If you are a end user, how does your provider deliver the calls to you?
The reason I ask is I read and hear a lot of issues where people are getting dropped calls or service being completely down because
the end user is receiving their SIP connections over the internet, the local cable/dsl provider. What should an end user look for in a ISP/ITSP when choosing a VoIP provider in regards to the most reliable service?