Hi,
I am currently evaluating asterisk 16. I have noticed an issue using
application playback. The beginning and the end of the audio file are
missing. If I use answer and wait(1) before playback, the beginning is
correct. I am using chan_sip, if this is of interest.
Best regards
Karsten
--
I have a FreePBX system with PRI trunks that's doing a number of things very
nicely, but frustrating me in one area.
I am using a Grandstream GXW-4008 in an off-premises location to provide
POTS service on four ports (this device worked fine in an early
application using a hardware VPN to the
For those who are interested, the problem appears to NOT exist in 1.2Beta2.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237
I have cross-posted this all over the place, and sent a copy directly to digium
support, in the hope of getting to the bottom of a problem that has me pulling
my hair out.
I currently have 2 production PSTN gateway servers, running asterisk 1.2beta and
TE406P cards (upgraded 405 cards, with
