Re: [SR-Users] Retransmission handling

2017-10-19 Thread Volkan Oransoy 
It seems I was digging the wrong page. Documentation is pretty clear.
Thank you guys. 

/Volkan



> On 19 Oct 2017, at 20:22, Daniel-Constantin Mierla  wrote:
> 
> 
> 
> On 19.10.17 17:41, Alex Balashov wrote:
>> On Thu, Oct 19, 2017 at 11:06:21AM +0300, Volkan Oransoy wrote:
>> 
>>> I wonder purpose of this block in default script. What I didn’t understand 
>>> is the purpose of "t_check_trans()”. Accoording to documentation, this 
>>> method returns a boolean. So why do we use it even we don’t check output?
>>> 
>>> # handle retransmissions
>>> if (!is_method("ACK")) {
>>> if(t_precheck_trans()) {
>>> t_check_trans();
>>> exit;
>>> }
>>> t_check_trans();
>>> }
>> Because t_check_trans() has certain behaviours simply in virtue of being
>> called:
>> 
>> https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.f.t_check_trans
>> 
>> For example:
>> 
>>   "For ACKs to negative replies or for ACKs to local transactions it will
>>terminate the script if the ACK belongs to a transaction"
>> 
>> 
>> 
> Just to highlight, the other section in the docs is also relevant,
> respectively stopping the config execution if it was a retransmission
> after sending back the last response.
> 
> Cheers,
> Daniel
> 
> -- 
> Daniel-Constantin Mierla
> www.twitter.com/miconda  -- 
> www.linkedin.com/in/miconda 
> Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com 
> 
> Kamailio World Conference - www.kamailioworld.com 
> 
> 
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Re: [SR-Users] Retransmission handling

2017-10-19 Thread Daniel-Constantin Mierla 


On 19.10.17 17:41, Alex Balashov wrote:
> On Thu, Oct 19, 2017 at 11:06:21AM +0300, Volkan Oransoy wrote:
>
>> I wonder purpose of this block in default script. What I didn’t understand 
>> is the purpose of "t_check_trans()”. Accoording to documentation, this 
>> method returns a boolean. So why do we use it even we don’t check output?
>>
>>  # handle retransmissions
>>  if (!is_method("ACK")) {
>>  if(t_precheck_trans()) {
>>  t_check_trans();
>>  exit;
>>  }
>>  t_check_trans();
>>  }
> Because t_check_trans() has certain behaviours simply in virtue of being
> called:
>
> https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.f.t_check_trans
>
> For example:
>
>"For ACKs to negative replies or for ACKs to local transactions it will
> terminate the script if the ACK belongs to a transaction"
>
>
>
Just to highlight, the other section in the docs is also relevant,
respectively stopping the config execution if it was a retransmission
after sending back the last response.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com
Kamailio World Conference - www.kamailioworld.com


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Re: [SR-Users] Retransmission handling

2017-10-19 Thread Alex Balashov 
On Thu, Oct 19, 2017 at 11:06:21AM +0300, Volkan Oransoy wrote:

> I wonder purpose of this block in default script. What I didn’t understand is 
> the purpose of "t_check_trans()”. Accoording to documentation, this method 
> returns a boolean. So why do we use it even we don’t check output?
> 
>   # handle retransmissions
>   if (!is_method("ACK")) {
>   if(t_precheck_trans()) {
>   t_check_trans();
>   exit;
>   }
>   t_check_trans();
>   }

Because t_check_trans() has certain behaviours simply in virtue of being
called:

https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.f.t_check_trans

For example:

   "For ACKs to negative replies or for ACKs to local transactions it will
terminate the script if the ACK belongs to a transaction"

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [SR-Users] Kamailio and FreeSwitch integration

2017-10-19 Thread Timothy oladapo olawuyi 
Hello Sergey,

 

User(s) will need to calling-in and listen to announcement. Can rtpProxy handle 
that?

 

 

Regards

 

 

Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah 
Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef 
Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001,+2348098797928 
|Skype: daptims | htttp://www.pethahiah.com  

 

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Sergey 
Safarov
Sent: Thursday, October 19, 2017 12:24 PM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio and FreeSwitch integration

 

Why not use rtpProxy module for media?

 

Sergey

 

чт, 19 окт. 2017 г. в 14:14, Timothy oladapo olawuyi  >:



 

 

 

 

Dear All,

 

I am a new entrant in Kamailio though little understanding of SIP and 
FreeSwitch soft switch, however, I have a project to integrate Kamailio and 
Freeswitch.

Here is the scenario of what I want to implement;

Kamailio will act as our SIP control while Freeswitch will act media server for 
incoming calls. 

Freeswitch will send all outgoing calls to Kamailio for onward transfer to our 
SIP provider network.

No registration, presence, location Accounting, Authentication etc. are 
required.

I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and 
Kamailio integration sample config available at 
http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still 
unable to figure out how to go with the configuration.

I will appreciate any one who can provide guideline for achieving the above 
scenario.

 

 

 

Regards

 

Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept |Pethahiah 
Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza , Lateef 
Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile: +2348052612001 
 ,+2348098797928   
|Skype: daptims | htttp://www.pethahiah.com

 

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Re: [SR-Users] Kamailio and FreeSwitch integration

2017-10-19 Thread Daniel Tryba 
On Thu, Oct 19, 2017 at 12:13:47PM +0100, Timothy oladapo olawuyi wrote:
> Kamailio will act as our SIP control while Freeswitch will act media server
> for incoming calls. 
> 
> Freeswitch will send all outgoing calls to Kamailio for onward transfer to
> our SIP provider network.
> 
> No registration, presence, location Accounting, Authentication etc. are
> required.
> 
> I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and
> Kamailio integration sample config available at
> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still
> unable to figure out how to go with the configuration.

That sample does all the things you don't want to implement.
 
> I will appreciate any one who can provide guideline for achieving the above
> scenario.

Your required config is a simple proxy.
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher.ex.config
contains a config example, which sends all traffic to 1 dispatcher. You
need to modify this a little. Add a second dispatcher so you have 1 for
your upstream and 1 for your freeswitch. If a message arrives from 1
dispatcher send it to the other. For example something like:

route[DISPATCH] 
{
  if(ds_is_from_list("1"))
  {
ds_select_dst("2", "4");
  }
  else if(ds_is_from_list("2"))
  {
ds_select_dst("1", "4");
  }
  else
  {
sl_send_reply("403","Forbidden");
exit;
  }
...

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Re: [SR-Users] Kamailio and FreeSwitch integration

2017-10-19 Thread Sergey Safarov 
Why not use rtpProxy module for media?

Sergey

чт, 19 окт. 2017 г. в 14:14, Timothy oladapo olawuyi :

>
>
>
>
>
>
>
>
> Dear All,
>
>
>
> I am a new entrant in Kamailio though little understanding of SIP and
> FreeSwitch soft switch, however, I have a project to integrate Kamailio and
> Freeswitch.
>
> Here is the scenario of what I want to implement;
>
> Kamailio will act as our SIP control while Freeswitch will act media
> server for incoming calls.
>
> Freeswitch will send all outgoing calls to Kamailio for onward transfer to
> our SIP provider network.
>
> No registration, presence, location Accounting, Authentication etc. are
> required.
>
> I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and
> Kamailio integration sample config available at
> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am
> still unable to figure out how to go with the configuration.
>
> I will appreciate any one who can provide guideline for achieving the
> above scenario.
>
>
>
>
>
>
>
> Regards
>
>
>
> *Olawuyi Timothy Oladapo **|* *IT/VAS Operation** |** Information System
> Dept **|**Pethahiah Rehoboth INTL. Limited* | *2nd Floor Rubby Block All
> seasons Plaza , Lateef Jakande Rd, Agidingbi, Ikeja, Lagos.* | *Mobile:**
> +2348052612001 <+234%20805%20261%202001>,+2348098797928
> <+234%20809%20879%207928> *|*Skype*: daptims | htttp://www.pethahiah.com
>
>
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[SR-Users] Kamailio and FreeSwitch integration

2017-10-19 Thread Timothy oladapo olawuyi 


 

 

 

 

Dear All,

 

I am a new entrant in Kamailio though little understanding of SIP and
FreeSwitch soft switch, however, I have a project to integrate Kamailio and
Freeswitch.

Here is the scenario of what I want to implement;

Kamailio will act as our SIP control while Freeswitch will act media server
for incoming calls. 

Freeswitch will send all outgoing calls to Kamailio for onward transfer to
our SIP provider network.

No registration, presence, location Accounting, Authentication etc. are
required.

I have gone through the book SIP ROUTING WITH KAMAILIO and FreeSwitch and
Kamailio integration sample config available at
http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms am still
unable to figure out how to go with the configuration.

I will appreciate any one who can provide guideline for achieving the above
scenario.

 

 

 

Regards

 

Olawuyi Timothy Oladapo | IT/VAS Operation | Information System Dept
|Pethahiah Rehoboth INTL. Limited | 2nd Floor Rubby Block All seasons Plaza
, Lateef Jakande Rd, Agidingbi, Ikeja, Lagos. | Mobile:
+2348052612001,+2348098797928 |Skype: daptims |
 htttp://www.pethahiah.com

 

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[SR-Users] Retransmission handling

2017-10-19 Thread Volkan Oransoy 
Hi all,

I wonder purpose of this block in default script. What I didn’t understand is 
the purpose of "t_check_trans()”. Accoording to documentation, this method 
returns a boolean. So why do we use it even we don’t check output?

# handle retransmissions
if (!is_method("ACK")) {
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
}


/Volkan



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Re: [SR-Users] port forwarding

2017-10-19 Thread Vladyslav Zakhozhai 
Hi Anrzej,

Right question is 50% of solution. Unfortunately the question is wrong.
Almost nothing is clear.

You can register your phones (clients) and you even can start or receive
calls but there is an issue with audio/video? Is it correct?

In this case this problem is not related to kamailio. It is related to
pbx/media server that is possibly behind kamailio. But maybe there is no
any pbx behind?

Without additional configuration audio/video traffic will go end-to-end
which is a problem with external clients with no public IP.

And yes, you are right about 5060/udp. But everything depends on your
configuration which is not clear. Maybe you need additionaly 5060/tcp,
5061/tcp (TLS).

But ports range for media traffic depends on media server/pbx/rtp proxy
configuration and nobody can answer this question but you.

I hope this tips will help you a little bit.


On Tue, Oct 17, 2017 at 1:40 AM, Andrzej Kaczmarczyk 
wrote:

> Hello,
>
> This is probably trivial, but I'm quite new in kamailio and trying to
> handle it for my personal purposes (so - no trainings, no support etc :) )
> - for few (up to 20) users and rare connections, more to learn something
> than to have production system. No plans to go outside my own server.
>
> I have LAN. One of machines in this LAN is server with kamailio onboard.
> It works, but only with local hostname (user@hostname).
> Now I want to go outside (with clients).
>
> As temporarty solution I have only dynamic DNS (no-ip.org) and I didn't
> expect it working. But tried with IP adress (u...@wan.ip.add.ress).
>
> And... I don't know which ports to redirect on router.
> Now there are redirected 5060 and 1-3 (maybe it is not necessary
> but first I want it working) - taken from some on-line info.
>
> I can:
> - log in to all my accounts,
>
> - depends on client - call to another user and in some cases I can receive
> the call (so it is probably issue on my clients, because it is possible to
> make a call at all).
>
> But I cannot transmit any data: connections are completely silent in any
> case and I cannot start video - although cameras are findable.
>
>
>
> Which ports in fact are necessary?
>
> I want audio and video communication, presence info and probably nothing
> more. Which ports should I redirect?
>
> --
> Pozdrawiam
> Andrzej Kaczmarczyk
>
>
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-- 
С уважением,
Владислав Захожай
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