Re: [SR-Users] SIREMIS 4.3.0 and Kamilio 5.1.0-dev1

[David Villasmil]

Looks like you have the domain duplicated on the table?
On Fri, May 12, 2017 at 11:14 PM Emil Morganti 
wrote:

> Hi everybody,
> I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to
> the second step i get an error:
>
> ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate
> entry '127.0.0.1' for key 'domain_idx'
>
> Some solution?
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[SR-Users] SIREMIS 4.3.0 and Kamilio 5.1.0-dev1

[Emil Morganti]

Hi everybody,
I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to 
the second step i get an error:


ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate 
entry '127.0.0.1' for key 'domain_idx'


Some solution?
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Re: [SR-Users] TLS to TCP

[David Villasmil]

Hello,

Thanks for the replies guys!

Juha was right, it's the client... funny thing, though:

When calling:

Client->kamailio->asterisk->gw   This works fine...

But when calling:

Client->kamilio->freeswitch->gwThis does NOT work...

I'm thinking maybe there's some topology hiding somewhere, so that the
client doesn't realizes the siganlling is being downgraded...


Any ideas? (I will take a look at the other kam's config)

David

ᐧ

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337

On Fri, May 12, 2017 at 6:48 PM, Colin Morelli 
wrote:

> Kamailio could be ending the call, though it may also be one of the
> endpoints.
>
> Anyway, if your clients are dialing sips: URIs, then it is required that
> the signaling be TLS end-to-end. If you are trying to translate TLS to TCP,
> you should use sip:u...@domain.com;transport=tls. This should enforce TLS
> from the client -> proxy, but allow the proxy to use its preferred
> transport.
>
> The reason the call wouldn't end until it's established is because it's
> not until this time that the any party receives a list of Record-Route
> headers. If using sips: and a record-route comes back that indicates that a
> hop did not use TLS, the call would end.
>
> Best,
> Colin
>
> On Fri, May 12, 2017 at 12:44 PM, Juha Heinanen  wrote:
>
>> David Villasmil writes:
>>
>> > I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on
>> tcp,
>> > but AFTER the call is established, kamailio hangs the call with "SIPS
>> > required"...
>>
>> Are you sure that it is K that hangs the established call?
>>
>> -- Juha
>>
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Re: [SR-Users] TLS to TCP

[Colin Morelli]

Kamailio could be ending the call, though it may also be one of the
endpoints.

Anyway, if your clients are dialing sips: URIs, then it is required that
the signaling be TLS end-to-end. If you are trying to translate TLS to TCP,
you should use sip:u...@domain.com;transport=tls. This should enforce TLS
from the client -> proxy, but allow the proxy to use its preferred
transport.

The reason the call wouldn't end until it's established is because it's not
until this time that the any party receives a list of Record-Route headers.
If using sips: and a record-route comes back that indicates that a hop did
not use TLS, the call would end.

Best,
Colin

On Fri, May 12, 2017 at 12:44 PM, Juha Heinanen  wrote:

> David Villasmil writes:
>
> > I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on
> tcp,
> > but AFTER the call is established, kamailio hangs the call with "SIPS
> > required"...
>
> Are you sure that it is K that hangs the established call?
>
> -- Juha
>
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[SR-Users] TLS to TCP

[Juha Heinanen]

David Villasmil writes:

> I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp,
> but AFTER the call is established, kamailio hangs the call with "SIPS
> required"...

Are you sure that it is K that hangs the established call?

-- Juha

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[SR-Users] TLS to TCP

[David Villasmil]

Hello Guys,

I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp,
but AFTER the call is established, kamailio hangs the call with "SIPS
required"...

Has this happened to anyone?

Regards,

David Villasmil
email: david.villasmil.w...@gmail.com
phone: +34669448337
ᐧ
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Re: [SR-Users] kamcmd domain.dump ERROR: reply too big

[Daniel-Constantin Mierla]

Hello,



On 12.05.17 15:54, Denys Pozniak wrote:
> Hello!
>
> I have updated Kamailio to version 5.1.0-dev2 and started to obtain
> next error:
>
> /kamcmd> domain.dump/
> /ERROR: reply too big/
>
> Tried to increase ctl buffer size, but result is the same.
>
> /loadmodule "ctl.so"/
> /modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")/
> /modparam("ctl", "binrpc_max_body_size", 32000)/
> /modparam("ctl", "binrpc_struct_max_body_size", 3200)/
/Is in the above line missing a 0 than what you wanted?

I also increased the buffers in kamcmd code, a commit was just pushed in
git.

Cheers,
Daniel

/
>
> Please point me where to find issue.
>
>
>
> /# uname -a/
> /Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017
> x86_64 x86_64 x86_64 GNU/Linux/
> /
> /
> /# cat /etc/lsb-release/
> /DISTRIB_ID=Ubuntu/
> /DISTRIB_RELEASE=16.04/
> /DISTRIB_CODENAME=xenial/
> /DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"/
> /
> /
>
> BR,
> Denys
>
>
>
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[SR-Users] kamcmd domain.dump ERROR: reply too big

[Denys Pozniak]

Hello!

I have updated Kamailio to version 5.1.0-dev2 and started to obtain next
error:

*kamcmd> domain.dump*
*ERROR: reply too big*

Tried to increase ctl buffer size, but result is the same.

*loadmodule "ctl.so"*
*modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")*
*modparam("ctl", "binrpc_max_body_size", 32000)*
*modparam("ctl", "binrpc_struct_max_body_size", 3200)*

Please point me where to find issue.



*# uname -a*
*Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017 x86_64
x86_64 x86_64 GNU/Linux*

*# cat /etc/lsb-release*
*DISTRIB_ID=Ubuntu*
*DISTRIB_RELEASE=16.04*
*DISTRIB_CODENAME=xenial*
*DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"*


BR,
Denys
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Re: [SR-Users] t_drop_replies not working with t_suspend in failure route

[Huber Andreas]

Hi,
We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon.
Cool so it will be fixed when we migrate !

Thanks,
Andreas

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of 
Federico Cabiddu
Sent: vendredi 12 mai 2017 11:56
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure 
route

Hi,
which version are you using?
A similar case had been reported some months ago and it should be fixed in 5.0.

Regards,

Federico

On Fri, May 12, 2017 at 11:44 AM, Huber Andreas 
> wrote:
Hello,

We have a use case where we suspend a transaction in a  failure_route to give 
UEs that might be woken by a push notification more time to REGISTER and join 
the INVITE.
We’d like to drop the previous branches in this case. I tried using 
t_drop_replies() but it has no effect.

The doc states that t_drop_replies() is only working if a new branch is added. 
And from my understanding t_suspend() adds a new branch.
But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am 
I missing something?

Kind Regards,
Andreas



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Re: [SR-Users] t_drop_replies not working with t_suspend in failure route

[Federico Cabiddu]

Hi,
which version are you using?
A similar case had been reported some months ago and it should be fixed in
5.0.

Regards,

Federico

On Fri, May 12, 2017 at 11:44 AM, Huber Andreas 
wrote:

> Hello,
>
>
>
> We have a use case where we suspend a transaction in a  failure_route to
> give UEs that might be woken by a push notification more time to REGISTER
> and join the INVITE.
>
> We’d like to drop the previous branches in this case. I tried using
> t_drop_replies() but it has no effect.
>
>
>
> The doc states that t_drop_replies() is only working if a new branch is
> added. And from my understanding t_suspend() adds a new branch.
>
> But is it possible that t_drop_replies() cannot be used with t_suspend()?
> Or am I missing something?
>
>
>
> Kind Regards,
>
> Andreas
>
>
>
>
>
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[SR-Users] t_drop_replies not working with t_suspend in failure route

[Huber Andreas]

Hello,

We have a use case where we suspend a transaction in a  failure_route to give 
UEs that might be woken by a push notification more time to REGISTER and join 
the INVITE.
We'd like to drop the previous branches in this case. I tried using 
t_drop_replies() but it has no effect.

The doc states that t_drop_replies() is only working if a new branch is added. 
And from my understanding t_suspend() adds a new branch.
But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am 
I missing something?

Kind Regards,
Andreas


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Re: [SR-Users] Can rtpproxy stream audio inside a call? (Maxim Sobolev)

[Yufei Tao]

Hi,

Thank you very much for the answers! I'll have a play.

Cheers,
Yufei

Message: 9
Date: Mon, 8 May 2017 01:01:51 -0700
From: Maxim Sobolev 
To: sr-users@lists.kamailio.org, Daniel-Constantin Mierla

Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call?
Message-ID:

Content-Type: text/plain; charset="utf-8"

Daniel is right. Technically speaking it can be done at any point after the
media session has been established, but you need some kind of trigger to
start/stop it from your routing script. Such as re-INVITE for example.

-Max

On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" 
wrote:

> Hello,
>
> it can play rtp files encoded in the format expected by the audio codec. I
> used only when there was a re-invite (like putting the call on hold).
>
> Cheers,
> Daniel
>
> On 04.05.17 17:29, Yufei Tao wrote:
>
> Hi,
>
> Just a quick question: if I use the Kamailio rtpproxy module with the
Sippy
> RTPproxy, can I stream audio within a call? Or is it only possible before
> call audio is established?
>
> For example can I mix/playback audio file into an existing established
> call session at any time using functions rtpproxy_stream2uac() and
rtpproxy_stream2uas()
> somehow? If it is possible, does it need to be triggered by an in-dialog
> message from either UAC or UAS?
>
> Cheers,
> Yufei
>
>
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