[OpenSIPS-Users] B2BUA marketting scenario

Răzvan Crainea razvan at opensips.org
Wed Dec 9 17:23:50 CET 2015


Hi, Sebastian!

Thanks for reporting this bug. I did manage to replicate this problem 
and found the issue in the code. I've just pushed a fix on 1.11, 2.1 and 
the master branches.
Could you please update your sources and run a test again?

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/01/2015 11:23 AM, Bogdan-Andrei Iancu wrote:
> Hi Sebastian,
>
> Not yet, but I'm preparing the setup to run the test and fix. Anyhow, 
> I haven't forgot about this !
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://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 
>> <sastre.sebastian at gmail.com <mailto:sastre.sebastian at gmail.com>> 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
>>     <bogdan at opensips.org <mailto:bogdan at 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 Developer
>>         http://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
>>>         <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>>
>>>             Hi Sebastian,
>>>
>>>             You mentioned yesterday on IRC channel that you fixed
>>>             the problem ?
>>>
>>>             Regards,
>>>
>>>             Bogdan-Andrei Iancu
>>>             OpenSIPS Founder and Developer
>>>             http://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 at 172.10.1.107:5060
>>>>             <http://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 at 172.10.1.20:5060 <http://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 at 73.139.116.217
>>>>             <mailto:sip%3Asebas3 at 73.139.116.217>
>>>>             172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
>>>>             sip:1 at 172.10.1.20:5060;transport=udp
>>>>             <mailto:sip:1 at 172.10.1.20:5060;transport=udp>
>>>>
>>>>             172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>>>>             sip:DialerProxy at 172.10.1.21:5060
>>>>             <http://sip:DialerProxy@172.10.1.21:5060>
>>>>             172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
>>>>             sip:1 at 172.10.1.20:5060;transport=udp
>>>>             <mailto:sip:1 at 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
>>>>             <bogdan at opensips.org <mailto:bogdan at 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 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>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>                 _______________________________________________
>>>>>                 Users mailing list
>>>>>                 Users at lists.opensips.org
>>>>>                 <mailto:Users at lists.opensips.org>
>>>>>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>
>>
>>
>
>
>
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