Hi Sebastian,
Razvan uploaded a fix for this problem - please check it.
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 31.08.2015 23:38, Sebastian Sastre wrote:
Bodgan,
Did you have a change to look into this? just curious to know if you
replicated the problem.
thanks !
On Tue, Aug 18, 2015 at 12:16 PM, Sebastian Sastre
<[email protected] <mailto:[email protected]>> wrote:
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
<[email protected] <mailto:[email protected]>> 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 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
<[email protected] <mailto:[email protected]>> 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:[email protected]:5060
<http://sip:[email protected]: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:[email protected]:5060 <http://sip:[email protected]: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:[email protected]
<mailto:sip%[email protected]>
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:[email protected]:5060;transport=udp
<mailto:sip:[email protected]:5060;transport=udp>
172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:[email protected]:5060
<http://sip:[email protected]:5060>
172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:[email protected]:5060;transport=udp
<mailto:sip:[email protected]: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
<[email protected] <mailto:[email protected]>> 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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