I wonder if anyone else is having these problems. We are running
Asterisk 1.2.17, with an assortment of SIP users and peers. This is
running on an 600 MHz P3 with CentOS 4.4, and worked properly in
Asterisk 1.2.15. Nothing else running on the server except the usual
support stuff like sshd, a mostly idle httpd, and no GUI.
app_dictate works fine for recording, but on some calls during playback
the audio jumps around, playing fragments of the file. Using the fast
playback mode sometimes works, sometimes causes the jumping around to
get worse.
Incoming calls to the Dictate() application from different SIP carriers
and different hard and soft phones give drastically different results.
For instance, dialing in via an 01 Communications DID (resold by
Broadvoice) at 831-713-4569 fails on playback (as described, just
fragments of audio) every time. Dialing in via a Broadwing DID (resold
by Vitelity) at 831-621-1913 works. Calling from a Grandstream phone
fails, from a Cisco 7960 works most of the time, from a Motorola VT-1005
ATA always works.
All other playback modes including MOH work fine.
I have some clue, but not enough. Any ideas?
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