[OpenSIPS-Users] B2BUA marketting scenario
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Aug 18 13:42:48 CEST 2015
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
>
> 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
> 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>
>>
>>
>>
>>
>>
>>
>>
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>
>
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