On Sun, Apr 29, 2007 at 11:55:43AM +0100, Tim Panton wrote:
TP Theoretically Skype _can_ produce better audio quality than asterisk
TP as it supports a wideband codec
Asterisk 1.4 support G.722 transit, and trunk support G.722 transcoding.
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On Thu, May 25, 2006 at 04:42:57PM -0600, Colin Anderson wrote:
CA I looked long and hard at the LAN and it was basically narrowed down to the
CA switches. In this smaller install, several cheapo Dlink ($30) switches
What switches you mean? How they named?
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IAX2 trunking don't work with slinear codec. It's sounds like byteswapping
bug. When I change codec to alaw it works fine. When I disable trunking
it also works.
first server -- Opteron (32 bit mode)
second server -- Athlon
At first server it reproduced, e.g. with Playsound. At second server I
I have asterisk installation, using CVS-HEAD.
When call from D-Link 2004S (SIP) to _some_ numbers (e.g. mobile
numbers), after answer on other side D-Link user continue hear progress
tones.
Callee can hear, what D-Link user say, and ifter he try to send DTMF
(press any key on mobile phone),
I have asterisk installation, using CVS-HEAD.
When call from D-Link 2004S (SIP) to _some_ numbers (e.g. mobile
numbers), after answer on other side D-Link user continue hear progress
tones.
Callee can hear, what D-Link user say, and ifter he try to send DTMF
(press any key on mobile phone),
