Bodgan ! Thank you for the follow up ! i will test it and get back to you !
thanks again ! This is great news. On Fri, Dec 11, 2015 at 5:54 AM, Bogdan-Andrei Iancu <[email protected]> wrote: > Hi Sebastian, > > Razvan uploaded a fix for this problem - please check it. > > Thanks and regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > On 31.08.2015 23:38, Sebastian Sastre wrote: > > Bodgan, > > Did you have a change to look into this? just curious to know if you > replicated the problem. > > thanks ! > > > On Tue, Aug 18, 2015 at 12:16 PM, Sebastian Sastre < > <[email protected]>[email protected]> wrote: > >> 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 < >> <[email protected]>[email protected]> 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 < >>> <[email protected]>[email protected]> 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 >>>> <http://sip:[email protected]:5060>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 >>>> <http://sip:[email protected]:5060>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]>sip:[email protected] >>>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK >>>> <sip:[email protected]:5060;transport=udp> >>>> sip:[email protected]:5060;transport=udp >>>> >>>> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE >>>> <http://sip:[email protected]:5060> >>>> sip:[email protected]:5060 >>>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE >>>> <sip:[email protected]:5060;transport=udp> >>>> 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]>[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> >>>>> 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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