[OpenSIPS-Users] B2BUA marketting scenario

Sebastian Sastre sastre.sebastian at gmail.com
Tue Aug 18 15:20:27 CEST 2015


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
>>
>>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20150818/9344097d/attachment.htm>


More information about the Users mailing list