Hello and thanks for replying!
Steve,
The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
Actully ethereal,
It is encouraging to hear that
Actully ethereal
OK...
Try canreinvite=yes in the [general] section; this makes it the
default setting for all peers unless specified otherwise. Do the same
for nat=no in [general] to rule out all NAT'ing related issues. You
don't have tT in your Dial() statement, that's good. You say you
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do asterisk hangs on to the media path, how
in the
please turn on all the debug, warning, error etc messages in the
console, see logger.conf, then type sip peer peer1 debug and sip
peer peer2 debug to see the SIP messages.
How are you testing if asterisk is in the media path?
Regards
On 1/23/06, Steve Gladden [EMAIL PROTECTED] wrote:
been
Steve,
The mission is to actually get a reinvite to work on the lan.
There isn't anything special to get this working... normally. I trust
you verified the traffic flow with a network monitor tool (tcpdump?),
correct? Does SIP debug give you any info (i.e., does it match the
right peer) -- you
How are you testing if asterisk is in the media path?
Two ways:
One phone on a hub with ethereal on a laptop and watching the rtp
packets, pretty obvious that asterisk is staying in the media path.
and that the rtp i not coming from the other phone.
Way two, in the middle of an
On Mon, 23 Jan 2006, Steve Gladden wrote:
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do
