[OpenSIPS-Users] B2BUA marketting scenario

[Sebastian Sastre]

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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