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alpha16
2010 Feb 25
1
Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely
...al net side
to the phone. But I am unable to figure how make it do that.
The Asterisk log shows this.
[Feb 25 11:06:30] VERBOSE[1449] logger.c: --
Executing [s at macro-dialout-trunk:19]
^[[1;36;40mDial^[[0;37;40m("^[[1;35;40mSIP/2005-19dc0db8^[[0;37;40m",
"^[[1;35;40mSIP/ATT-alpi016-IPFlex1/19084611234,300,^[[0;37;40m") in new
stack
[Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP
RTP TOS bits 184
[Feb 25 11:06:30] VERBOSE[1449] logger.c: == Using SIP
RTP CoS mark 5
[Feb 25 11:06:30] VERBOSE[1449] logger.c: -- Called
ATT-alpi016-IPFlex1/19084611234
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