Displaying 1 result from an estimated 1 matches for "calldebug".
2007 Oct 01
1
Odd one way RTP on SIP to SIP calls
...k doesn't seem to send any RTP.
As far as I can tell, there isn't anything wrong with the call setup.
show core version shows:
Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
2007-05-17 06:39:34 UTC
SIP and RTP debugging on Asterisk shows this:
http://www.arnarson.net/~orn/calldebug.txt
On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
root @ build.trixbox.org on a i686 running Linux on 2007-04-25
19:59:21 UTC) on the same network (same subnet and physical location)
as the 1.4.4 this problem does not exist. There is no RTP problem when
SIP clients register...