Hello,
On 9/14/11 6:13 PM, CaptWho wrote:
Thanks Timo, I'll look into that, just have to figure out what you just said.
:confused: I'm at the bottom of the learning curve and I got dumped into
this after the guy that was dealing with it disappeared.
it will help also to watch the sip signaling traffic, using ngrep or
wireshark.
For example with ngrep, on the same server with kamailio:
ngrep -d any -qt -W byline port 5060
will capture all the incoming and outgoing sip messages. You can see if
there are some retransmissions, where the messages are sent, thus being
able to detect if there is anything wrong in routing. If you do nat
traversal, then also sdp and some headers must be updated in the proxy
server.
Cheers,
Daniel
Timo Klecker wrote:
Hello,
the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is
not send E2E but using the Kamailio. In this case you would have to
redirect
it.
In the content of INVITE and 200, 183 or ACK you should be able to see the
IPs for RTP data. Compare these with your phones.
O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX
Be sure, the RTP ports are open on your firewall, too. You can see the
ports
in the content, too.
M=audio XXXXX RTP/AVP 8 101
Greetings
Timo
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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