Bogdan, it appears to be broken as of 1.11 and 2.1 yes. I couldn't find any more indications in the logs that would point to a visible error, but the ACK still has no SDP.
I have a few machines to test this out with the different versions, let me know if you want a specific trace or core dump, happy to help. thanks ! On Tue, Aug 18, 2015 at 9:22 AM, Bogdan-Andrei Iancu <bog...@opensips.org> wrote: > Sebastian, > > So 1.11 and above are broken in this late ACK generation ? If so, I will > dig into . > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 18.08.2015 16:20, Sebastian Sastre wrote: > > Bodgan, > > Yes , i tried 1.11 and had the same issue, so i went down to 1.8 TLS and > it worked right away with the same scenario. A fee config changes but > overal its the standrad script. > > With 1.8 i see the sdp on the Ack and the call connects without problems. > Even video. > > Not sure why it did not work on higher versions. > > Regards, > > > On Tue, Aug 18, 2015 at 7:42 AM, Bogdan-Andrei Iancu <bog...@opensips.org> > wrote: > >> Hi Sebastian, >> >> You mentioned yesterday on IRC channel that you fixed the problem ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com >> >> On 17.08.2015 13:40, Sebastian Sastre wrote: >> >> 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:sebas3@172.10.1.107:5060 >> 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:1@172.10.1.20:5060, 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:sebas3@73.139.116.217 >> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK >> sip:1@172.10.1.20:5060;transport=udp >> >> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE >> sip:DialerProxy@172.10.1.21:5060 >> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE >> sip:1@172.10.1.20:5060;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 <bog...@opensips.org >> > 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 >>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> > >
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