subject:"\[OpenSIPS\-Users\] B2BUA marketting scenario"

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-12-11 Thread Bogdan-Andrei Iancu


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 
mailto:sastre.sebast...@gmail.com>> 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
mailto:bog...@opensips.org>> 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
mailto:bog...@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@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@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@73.139.116.217

172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060

172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:1@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
mailto:bog...@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

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-12-11 Thread Răzvan Crainea


Hi, Sebastian!

Thanks for reporting this bug. I did manage to replicate this problem 
and found the issue in the code. I've just pushed a fix on 1.11, 2.1 and 
the master branches.

Could you please update your sources and run a test again?

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/01/2015 11:23 AM, Bogdan-Andrei Iancu wrote:

Hi Sebastian,

Not yet, but I'm preparing the setup to run the test and fix. Anyhow, 
I haven't forgot about this !


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 
mailto:sastre.sebast...@gmail.com>> 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
mailto:bog...@opensips.org>> 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
mailto:bog...@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@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@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@73.139.116.217

172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060

172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:1@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
mailto:bog...@opensips.org>> wrote:

Hi Sebastian,

The 200OK from FS must be followed by ACK+

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-12-11 Thread Sebastian Sastre

Bodgan ! Thank you for the follow up !

i will test it and get back to you !

thanks again ! This is great news.


On Fri, Dec 11, 2015 at 5:54 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Sebastian,
>
> Razvan uploaded a fix for this problem - please check it.
>
> Thanks and regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://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 <
> sastre.sebast...@gmail.com> 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 <
>> bog...@opensips.org> 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 Developerhttp://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 <
>>> bog...@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@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@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@73.139.116.217
 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
 
 sip:1@172.10.1.20:5060;transport=udp

 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
 
 sip:DialerProxy@172.10.1.21:5060
 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
 
 sip:1@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 <
 bog...@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

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-09-07 Thread Bogdan-Andrei Iancu


Hi Sebastian,

Not yet, but I'm preparing the setup to run the test and fix. Anyhow, I 
haven't forgot about this !


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 
mailto:sastre.sebast...@gmail.com>> 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
mailto:bog...@opensips.org>> 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
mailto:bog...@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@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@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@73.139.116.217

172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060

172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:1@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
mailto:bog...@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:

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-31 Thread Sebastian Sastre

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 <
sastre.sebast...@gmail.com> 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 
> 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 Developerhttp://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 > > 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@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@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@73.139.116.217
>>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
>>> sip:1@172.10.1.20:5060;transport=udp
>>>
>>> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>>> sip:DialerProxy@172.10.1.21:5060
>>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
>>> sip:1@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 <
>>> bog...@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.

 
 >>> type="extern">
   
 
 
 client1
 
1
 
 
>>>

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-18 Thread Sebastian Sastre

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 
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 Developerhttp://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 
> 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@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@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@73.139.116.217
>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
>> sip:1@172.10.1.20:5060;transport=udp
>>
>> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>> sip:DialerProxy@172.10.1.21:5060
>> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
>> sip:1@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 > > 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.
>>>
>>> 
>>> >> type="extern">
>>>   
>>> 
>>> 
>>> client1
>>> 
>>>1
>>> 
>>> 
>>> 
>>> client2
>>> 
>>>2
>>> 
>>> 
>>> 
>>> 1
>>>   
>>> 
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>
>
___
Users mailing list
Us

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-18 Thread Sebastian Sastre

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 
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@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@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@73.139.116.217
> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
> sip:1@172.10.1.20:5060;transport=udp
>
> 172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
> sip:DialerProxy@172.10.1.21:5060
> 172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
> sip:1@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 
> 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.
>>
>> 
>> > type="extern">
>>   
>> 
>> 
>> client1
>> 
>>1
>> 
>> 
>> 
>> client2
>> 
>>2
>> 
>> 
>> 
>> 1
>>   
>> 
>>
>>
>>
>>
>>
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-18 Thread Bogdan-Andrei Iancu


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 
mailto:bog...@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@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@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@73.139.116.217 
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060

172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
sip:1@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
mailto:bog...@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.






client1

   1



client2

   2



1









___
Users mailing list
Users@lists.opensips.org  
http://lists.opensips.org/cgi-bin/mailman/listinfo/users








___
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Users@lists.ope

Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-18 Thread Bogdan-Andrei Iancu


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@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@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@73.139.116.217 
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK 
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE 
sip:DialerProxy@172.10.1.21:5060 
172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE 
sip:1@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 
mailto:bog...@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.



  


client1

   1



client2

   2



1
  








___
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Users@lists.opensips.org  
http://lists.opensips.org/cgi-bin/mailman/listinfo/users





___
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Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-17 Thread Bogdan-Andrei Iancu


Hi Sebastian,

I will put together a small setup and see what is going wrong.

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@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@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@73.139.116.217 
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK 
sip:1@172.10.1.20:5060;transport=udp


172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE 
sip:DialerProxy@172.10.1.21:5060 
172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE 
sip:1@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 
mailto:bog...@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.



  


client1

   1



client2

   2



1
  








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Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-17 Thread 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@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@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@73.139.116.217
172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK sip:1@172.10.1.20:5060
;transport=udp

172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
sip:DialerProxy@172.10.1.21:5060
172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE sip:1@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 
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.
>
> 
> 
>   
> 
> 
> client1
> 
>1
> 
> 
> 
> client2
> 
>2
> 
> 
> 
> 1
>   
> 
>
>
>
>
>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] B2BUA marketting scenario

2015-08-17 Thread Bogdan-Andrei Iancu


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.


type="extern">

  


client1

   1



client2

   2



1
  








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Users mailing list
Users@lists.opensips.org
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[OpenSIPS-Users] B2BUA marketting scenario

2015-08-16 Thread Sebastian Sastre

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.



  


client1

   1



client2

   2



1
  

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Users mailing list
Users@lists.opensips.org
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Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

Re: [OpenSIPS-Users] B2BUA marketting scenario

[OpenSIPS-Users] B2BUA marketting scenario

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