Hi Sebastian,
The 200OK from FS must be followed by ACK+SDP to linphone. See:
http://www.opensips.org/Documentation/Tutorials-B2BUA#toc14
If this does not happen -> do you see any errors in the logs (around the
processing of 200OK from FS) ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.08.2015 04:18, Sebastian Sastre wrote:
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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