[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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