[OpenSIPS-Users] B2BUA marketting scenario

Sebastian Sastre sastre.sebastian at gmail.com
Fri Dec 11 16:25:16 CET 2015


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 <bogdan at opensips.org>
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 <
> <sastre.sebastian at gmail.com>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>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 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 <
>>> <bogdan at opensips.org>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 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:sebas3@172.10.1.107:5060>sip:sebas3 at 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
>>>> <http://sip:1@172.10.1.20:5060>sip:1 at 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%3Asebas3 at 73.139.116.217>sip:sebas3 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>
>>>> sip:1 at 172.10.1.20:5060;transport=udp
>>>>
>>>> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>>>> <http://sip:DialerProxy@172.10.1.21:5060>
>>>> sip:DialerProxy at 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>
>>>> 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>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>
>>>>> 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 at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>
>>>
>>
>
>
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