This attached picture shows my lab setup. The Asterisk server has two network interfaces and it can simultaneously connected with both networks. I am calling from SIP Phone 1 to SIP Phone 2. The call establishes fine on SIP. But once the call is established there is no RTP packets exchange between two SIP phones. Can anybody guide me that what is it I am doing wrong.
regards
Set directmedia to no, or provide routing between the two sub-networks, and ensure that both phones no about the router If that doesn’t help, you will need to provide your configuration and traces of the SDP exchange in the call setup.
Note that many cut and paste examples show directmedia under an obsolete name of canreinvite.
I have already tried setting “directmedia = no” but the problem persists. Since Sip session is established without any problem, therefore the routing scheme seems fine, however no RTP packets are being exchanged somehow.
You will need to provide the configuration and SIP traces.
[1111]
type=friend
context=“Outgoing”
callerid=”1111”<1111>
host=dynamic
directmedia=no
disallow=all
allow=ulaw
allow=alaw
These are some of the parameters of sip.conf for sip phone 1. Right now I do not have sip trace as I am away. I can share more details on Monday.