Hi guys,
Im using the B2BUA module to send a call out to our subscribers and bridge
them with our IVR server on answer.
The subscriber side uses linphone and the media server is a freeswitch 1.6.
When placing the call thru the trigger scenario MI command, the initial
invite does not have any SDP inside which makes sense.
Once the 200ok is received from the linphone client, opensips uses the SDP
contained in the 200 to generate an invite to the freeswitch box. which is
great.
However, when the 200ok is received from freeswitch, the following ACK back
the linphone client does not contain the SDP and Linphone complains with
"No codec intersection" and sends an immediate bye.
Am i right to think that the sdp should go in the ack to create a late
offer?
Should i be sending a re invite?
any help appreciated.
My scenario is simple.
<?xml version="1.0"?>
<scenario id="dialer" name="MS start conditional" param="2" type="extern">
<init>
<bridge>
<client>
<id>client1</id>
<destination>
<value type="param">1</value>
</destination>
</client>
<client>
<id>client2</id>
<destination>
<value type="param">2</value>
</destination>
</client>
</bridge>
<state>1</state>
</init>
</scenario>
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