Bodgan, Thanks i wasn't sure on the ack process. This is the log , the scenario is triggered by a httpd json call.
INFO:b2b_logic:b2bl_add_client: adding entity [0x7f718dfa7068]->[B2B.173.7331923] to tuple [0x7f718dfa0cd0]->[685.0] WARNING:b2b_logic:b2bl_delete_entity: entity [0x7f718dfa2d18]->[] not found for tuple [685.0] INFO:b2b_logic:b2bl_delete_entity: delete tuple [685.0], entity [] INFO:b2b_logic:b2bl_add_client: adding entity [0x7f718dfa4d28]->[B2B.173.5533781] to tuple [0x7f718dfa0cd0]->[685.0] INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [B2B.173.7331923] - [B2B.173.5533781] and the trace looks like this 172.10.1.21 -> 172.10.1.107 SIP 436 Request: INVITE sip:[email protected]:5060 172.10.1.107 -> 172.10.1.21 SIP 346 Status: 100 Giving a try 172.10.1.107 -> 172.10.1.21 SIP 456 Status: 180 Ringing 172.10.1.107 -> 172.10.1.21 SIP/SDP 1088 Status: 200 Ok, with session description 172.10.1.21 -> 172.10.1.20 SIP/SDP 843 Request: INVITE sip:[email protected]:5060, with session description 172.10.1.20 -> 172.10.1.21 SIP 390 Status: 100 Trying 172.10.1.20 -> 172.10.1.21 SIP/SDP 1252 Status: 200 OK, with session description 172.10.1.21 -> 172.10.1.107 SIP 526 Request: ACK sip:[email protected] 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK sip:[email protected]:5060 ;transport=udp 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE sip:[email protected]:5060 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE sip:[email protected]:5060 ;transport=udp 172.10.1.20 -> 172.10.1.21 SIP 545 Status: 200 OK 172.10.1.21 -> 172.10.1.107 SIP 629 Status: 200 OK thanks ! On Mon, Aug 17, 2015 at 5:47 AM, Bogdan-Andrei Iancu <[email protected]> wrote: > 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 Developerhttp://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> > > > > > > > > _______________________________________________ > Users mailing > [email protected]http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >
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