[OpenSIPS-Users] B2BUA marketting scenario
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Aug 18 15:22:27 CEST 2015
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
>>
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20150818/bddbef71/attachment-0001.htm>
More information about the Users
mailing list