On Mon, Jan 30, 2012 at 7:31 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/28/2012 10:22 AM, Din Assegaf wrote:
The error message is misleading; you are having this problem because the
'm' line in the SDP with the 'audio' offer has a port number of 0 (zero).,
which means
Hi All,
I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
But when making A Call from SIP Client, I got cli Warning ... and no call
has been made.
My Sip Client is using lib java peers client http://peers.sourceforge.net/
with standard codec PCMU/PCMA
[Jan 28 23:03:32]
Hi All,
I am currently building GSM Based trunk, for voice and sms.
I am compiling the newest 1.6 (Asterisk
1.6.2.22http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.22.tar.gz)
asterisk with --bluetooth support,
and also asterisk-addons for chan_mobile support,
my os
