Asterisk PJSIP with Kamailio dispatcher

[lemongrass]

My setup consists of the SIP Client 101 and 102 registered to Kamailio. Client 101 tries to call Client 102.

I have the setup where Kamailio handles the registrations and location tracking. Asterisk should act as a kind of Media Server without holding any registrations.

For this, I have oriented my Kamailio config to this article asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site]

My Kamailio route[LOCATION] features the following section to route all INVITES to Asterisk:

	if(is_method("INVITE") && (!route(FROMASTERISK))) {
		# if new call from out there - send to Asterisk
		# - non-INVITE request are routed directly by Kamailio
		# - traffic from Asterisk is routed also directy by Kamailio
		route(TOASTERISK);
		exit;
	}

This seems to work fine as sngrep shows

The dialplan consists of the following

[internal]
exten => _X.,1,NoOp(test)
same => n,Dial(PJSIP/kamailio/sip:${EXTEN}@192.168.0.201:5060)

endpoint config

[kamailio]
type=endpoint
transport=transport-udp-nat
aors=kamailio
rewrite_contact=no

[kamailio]
type=aor
contact=sip:192.168.0.201:5060

Now the issue is that the INVITE coming out of Asterisk is missing the SDP information

Any idea why this happens?

[lemongrass]

Second sngrep showing the missing SDP:

[jcolp]

You haven’t configured any codecs, and in older versions that will result in no media stream.

[lemongrass]

Thanks, this helped!