[OpenSIPS-Users] B2BUA marketting scenario
Sebastian Sastre
sastre.sebastian at gmail.com
Tue Aug 18 18:16:14 CEST 2015
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>
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>
> 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 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
>> 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: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
>>
>> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>> 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
>> 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
>> > 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 at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>
>
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